Remove VoEFile and things it uses.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/3013033002
Cr-Commit-Position: refs/heads/master@{#19885}
diff --git a/test/mock_voice_engine.h b/test/mock_voice_engine.h
index 8f8fdac..bd65b0b 100644
--- a/test/mock_voice_engine.h
+++ b/test/mock_voice_engine.h
@@ -134,55 +134,6 @@
MOCK_METHOD2(SetOpusMaxPlaybackRate, int(int channel, int frequency_hz));
MOCK_METHOD2(SetOpusDtx, int(int channel, bool enable_dtx));
- // VoEFile
- MOCK_METHOD7(StartPlayingFileLocally,
- int(int channel,
- const char fileNameUTF8[1024],
- bool loop,
- FileFormats format,
- float volumeScaling,
- int startPointMs,
- int stopPointMs));
- MOCK_METHOD6(StartPlayingFileLocally,
- int(int channel,
- InStream* stream,
- FileFormats format,
- float volumeScaling,
- int startPointMs,
- int stopPointMs));
- MOCK_METHOD1(StopPlayingFileLocally, int(int channel));
- MOCK_METHOD1(IsPlayingFileLocally, int(int channel));
- MOCK_METHOD6(StartPlayingFileAsMicrophone,
- int(int channel,
- const char fileNameUTF8[1024],
- bool loop,
- bool mixWithMicrophone,
- FileFormats format,
- float volumeScaling));
- MOCK_METHOD5(StartPlayingFileAsMicrophone,
- int(int channel,
- InStream* stream,
- bool mixWithMicrophone,
- FileFormats format,
- float volumeScaling));
- MOCK_METHOD1(StopPlayingFileAsMicrophone, int(int channel));
- MOCK_METHOD1(IsPlayingFileAsMicrophone, int(int channel));
- MOCK_METHOD4(StartRecordingPlayout,
- int(int channel,
- const char* fileNameUTF8,
- CodecInst* compression,
- int maxSizeBytes));
- MOCK_METHOD1(StopRecordingPlayout, int(int channel));
- MOCK_METHOD3(StartRecordingPlayout,
- int(int channel, OutStream* stream, CodecInst* compression));
- MOCK_METHOD3(StartRecordingMicrophone,
- int(const char* fileNameUTF8,
- CodecInst* compression,
- int maxSizeBytes));
- MOCK_METHOD2(StartRecordingMicrophone,
- int(OutStream* stream, CodecInst* compression));
- MOCK_METHOD0(StopRecordingMicrophone, int());
-
// VoENetwork
MOCK_METHOD2(RegisterExternalTransport,
int(int channel, Transport& transport));
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index b61f909..57a94e2 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -8,69 +8,6 @@
import("../webrtc.gni")
-rtc_static_library("audio_coder") {
- sources = [
- "coder.cc",
- "coder.h",
- ]
- deps = [
- "..:webrtc_common",
- "../api/audio_codecs:builtin_audio_decoder_factory",
- "../api/audio_codecs:builtin_audio_encoder_factory",
- "../modules:module_api",
- "../modules/audio_coding",
- "../modules/audio_coding:audio_format_conversion",
- "../modules/audio_coding:rent_a_codec",
- ]
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-}
-
-rtc_static_library("file_player") {
- sources = [
- "file_player.cc",
- "file_player.h",
- ]
- deps = [
- ":audio_coder",
- "..:webrtc_common",
- "../common_audio",
- "../modules:module_api",
- "../modules/media_file",
- "../rtc_base:rtc_base_approved",
- ]
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-}
-
-rtc_static_library("file_recorder") {
- sources = [
- "file_recorder.cc",
- "file_recorder.h",
- ]
- deps = [
- ":audio_coder",
- "..:webrtc_common",
- "../audio/utility:audio_frame_operations",
- "../common_audio",
- "../modules:module_api",
- "../modules/media_file:media_file",
- "../rtc_base:rtc_base_approved",
- "../system_wrappers",
- ]
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-}
-
rtc_static_library("voice_engine") {
sources = [
"channel.cc",
@@ -82,7 +19,6 @@
"include/voe_base.h",
"include/voe_codec.h",
"include/voe_errors.h",
- "include/voe_file.h",
"include/voe_network.h",
"include/voe_rtp_rtcp.h",
"monitor_module.h",
@@ -102,8 +38,6 @@
"voe_base_impl.h",
"voe_codec_impl.cc",
"voe_codec_impl.h",
- "voe_file_impl.cc",
- "voe_file_impl.h",
"voe_network_impl.cc",
"voe_network_impl.h",
"voe_rtp_rtcp_impl.cc",
@@ -132,8 +66,6 @@
]
deps = [
":audio_level",
- ":file_player",
- ":file_recorder",
"..:webrtc_common",
"../api:array_view",
"../api:audio_mixer_api",
@@ -182,7 +114,6 @@
if (rtc_include_tests) {
rtc_test("voice_engine_unittests") {
deps = [
- ":file_player",
":voice_engine",
"../common_audio",
"../modules:module_api",
@@ -211,7 +142,6 @@
sources = [
"channel_unittest.cc",
- "file_player_unittests.cc",
"transport_feedback_packet_loss_tracker_unittest.cc",
"utility_unittest.cc",
"voe_base_unittest.cc",
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index cc668f2..e5435d6 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -651,8 +651,6 @@
// Store speech type for dead-or-alive detection
_outputSpeechType = audioFrame->speech_type_;
- ChannelState::State state = channel_state_.Get();
-
{
// Pass the audio buffers to an optional sink callback, before applying
// scaling/panning, as that applies to the mix operation.
@@ -680,21 +678,6 @@
AudioFrameOperations::ScaleWithSat(output_gain, audioFrame);
}
- // Mix decoded PCM output with file if file mixing is enabled
- if (state.output_file_playing) {
- MixAudioWithFile(*audioFrame, audioFrame->sample_rate_hz_);
- muted = false; // We may have added non-zero samples.
- }
-
- // Record playout if enabled
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (_outputFileRecording && output_file_recorder_) {
- output_file_recorder_->RecordAudioToFile(*audioFrame);
- }
- }
-
// Measure audio level (0-9)
// TODO(henrik.lundin) Use the |muted| information here too.
// TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| (see
@@ -774,20 +757,7 @@
highestNeeded = receiveFrequency;
}
- // Special case, if we're playing a file on the playout side
- // we take that frequency into consideration as well
- // This is not needed on sending side, since the codec will
- // limit the spectrum anyway.
- if (channel_state_.Get().output_file_playing) {
- rtc::CritScope cs(&_fileCritSect);
- if (output_file_player_) {
- if (output_file_player_->Frequency() > highestNeeded) {
- highestNeeded = output_file_player_->Frequency();
- }
- }
- }
-
- return (highestNeeded);
+ return highestNeeded;
}
int32_t Channel::CreateChannel(Channel*& channel,
@@ -808,53 +778,6 @@
return 0;
}
-void Channel::PlayNotification(int32_t id, uint32_t durationMs) {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::PlayNotification(id=%d, durationMs=%d)", id,
- durationMs);
-
- // Not implement yet
-}
-
-void Channel::RecordNotification(int32_t id, uint32_t durationMs) {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::RecordNotification(id=%d, durationMs=%d)", id,
- durationMs);
-
- // Not implement yet
-}
-
-void Channel::PlayFileEnded(int32_t id) {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::PlayFileEnded(id=%d)", id);
-
- if (id == _inputFilePlayerId) {
- channel_state_.SetInputFilePlaying(false);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::PlayFileEnded() => input file player module is"
- " shutdown");
- } else if (id == _outputFilePlayerId) {
- channel_state_.SetOutputFilePlaying(false);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::PlayFileEnded() => output file player module is"
- " shutdown");
- }
-}
-
-void Channel::RecordFileEnded(int32_t id) {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::RecordFileEnded(id=%d)", id);
-
- assert(id == _outputFileRecorderId);
-
- rtc::CritScope cs(&_fileCritSect);
-
- _outputFileRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::RecordFileEnded() => output file recorder module is"
- " shutdown");
-}
-
Channel::Channel(int32_t channelId,
uint32_t instanceId,
const VoEBase::ChannelConfig& config)
@@ -874,12 +797,6 @@
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
_outputAudioLevel(),
_externalTransport(false),
- // Avoid conflict with other channels by adding 1024 - 1026,
- // won't use as much as 1024 channels.
- _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
- _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
- _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
- _outputFileRecording(false),
_timeStamp(0), // This is just an offset, RTP module will add it's own
// random offset
ntp_estimator_(Clock::GetRealTimeClock()),
@@ -899,7 +816,6 @@
input_mute_(false),
previous_frame_muted_(false),
_outputGain(1.0f),
- _mixFileWithMicrophone(false),
_includeAudioLevelIndication(false),
transport_overhead_per_packet_(0),
rtp_overhead_per_packet_(0),
@@ -1088,22 +1004,6 @@
StopSend();
StopPlayout();
- {
- rtc::CritScope cs(&_fileCritSect);
- if (input_file_player_) {
- input_file_player_->RegisterModuleFileCallback(NULL);
- input_file_player_->StopPlayingFile();
- }
- if (output_file_player_) {
- output_file_player_->RegisterModuleFileCallback(NULL);
- output_file_player_->StopPlayingFile();
- }
- if (output_file_recorder_) {
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_->StopRecording();
- }
- }
-
// The order to safely shutdown modules in a channel is:
// 1. De-register callbacks in modules
// 2. De-register modules in process thread
@@ -1174,8 +1074,6 @@
}
channel_state_.SetPlaying(true);
- if (RegisterFilePlayingToMixer() != 0)
- return -1;
return 0;
}
@@ -1823,508 +1721,6 @@
return 0;
}
-int Channel::StartPlayingFileLocally(const char* fileName,
- bool loop,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartPlayingFileLocally(fileNameUTF8[]=%s, loop=%d,"
- " format=%d, volumeScaling=%5.3f, startPosition=%d, "
- "stopPosition=%d)",
- fileName, loop, format, volumeScaling, startPosition,
- stopPosition);
-
- if (channel_state_.Get().output_file_playing) {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceError,
- "StartPlayingFileLocally() is already playing");
- return -1;
- }
-
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (output_file_player_) {
- output_file_player_->RegisterModuleFileCallback(NULL);
- output_file_player_.reset();
- }
-
- output_file_player_ = FilePlayer::CreateFilePlayer(
- _outputFilePlayerId, (const FileFormats)format);
-
- if (!output_file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileLocally() filePlayer format is not correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (output_file_player_->StartPlayingFile(
- fileName, loop, startPosition, volumeScaling, notificationTime,
- stopPosition, (const CodecInst*)codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- output_file_player_->StopPlayingFile();
- output_file_player_.reset();
- return -1;
- }
- output_file_player_->RegisterModuleFileCallback(this);
- channel_state_.SetOutputFilePlaying(true);
- }
-
- if (RegisterFilePlayingToMixer() != 0)
- return -1;
-
- return 0;
-}
-
-int Channel::StartPlayingFileLocally(InStream* stream,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartPlayingFileLocally(format=%d,"
- " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
- format, volumeScaling, startPosition, stopPosition);
-
- if (stream == NULL) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFileLocally() NULL as input stream");
- return -1;
- }
-
- if (channel_state_.Get().output_file_playing) {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceError,
- "StartPlayingFileLocally() is already playing");
- return -1;
- }
-
- {
- rtc::CritScope cs(&_fileCritSect);
-
- // Destroy the old instance
- if (output_file_player_) {
- output_file_player_->RegisterModuleFileCallback(NULL);
- output_file_player_.reset();
- }
-
- // Create the instance
- output_file_player_ = FilePlayer::CreateFilePlayer(
- _outputFilePlayerId, (const FileFormats)format);
-
- if (!output_file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileLocally() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (output_file_player_->StartPlayingFile(stream, startPosition,
- volumeScaling, notificationTime,
- stopPosition, codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to "
- "start file playout");
- output_file_player_->StopPlayingFile();
- output_file_player_.reset();
- return -1;
- }
- output_file_player_->RegisterModuleFileCallback(this);
- channel_state_.SetOutputFilePlaying(true);
- }
-
- if (RegisterFilePlayingToMixer() != 0)
- return -1;
-
- return 0;
-}
-
-int Channel::StopPlayingFileLocally() {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StopPlayingFileLocally()");
-
- if (!channel_state_.Get().output_file_playing) {
- return 0;
- }
-
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (output_file_player_->StopPlayingFile() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopPlayingFile() could not stop playing");
- return -1;
- }
- output_file_player_->RegisterModuleFileCallback(NULL);
- output_file_player_.reset();
- channel_state_.SetOutputFilePlaying(false);
- }
- // _fileCritSect cannot be taken while calling
- // SetAnonymousMixibilityStatus. Refer to comments in
- // StartPlayingFileLocally(const char* ...) for more details.
- if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, false) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StopPlayingFile() failed to stop participant from playing as"
- "file in the mixer");
- return -1;
- }
-
- return 0;
-}
-
-int Channel::IsPlayingFileLocally() const {
- return channel_state_.Get().output_file_playing;
-}
-
-int Channel::RegisterFilePlayingToMixer() {
- // Return success for not registering for file playing to mixer if:
- // 1. playing file before playout is started on that channel.
- // 2. starting playout without file playing on that channel.
- if (!channel_state_.Get().playing ||
- !channel_state_.Get().output_file_playing) {
- return 0;
- }
-
- // |_fileCritSect| cannot be taken while calling
- // SetAnonymousMixabilityStatus() since as soon as the participant is added
- // frames can be pulled by the mixer. Since the frames are generated from
- // the file, _fileCritSect will be taken. This would result in a deadlock.
- if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0) {
- channel_state_.SetOutputFilePlaying(false);
- rtc::CritScope cs(&_fileCritSect);
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StartPlayingFile() failed to add participant as file to mixer");
- output_file_player_->StopPlayingFile();
- output_file_player_.reset();
- return -1;
- }
-
- return 0;
-}
-
-int Channel::StartPlayingFileAsMicrophone(const char* fileName,
- bool loop,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartPlayingFileAsMicrophone(fileNameUTF8[]=%s, "
- "loop=%d, format=%d, volumeScaling=%5.3f, startPosition=%d, "
- "stopPosition=%d)",
- fileName, loop, format, volumeScaling, startPosition,
- stopPosition);
-
- rtc::CritScope cs(&_fileCritSect);
-
- if (channel_state_.Get().input_file_playing) {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() filePlayer is playing");
- return 0;
- }
-
- // Destroy the old instance
- if (input_file_player_) {
- input_file_player_->RegisterModuleFileCallback(NULL);
- input_file_player_.reset();
- }
-
- // Create the instance
- input_file_player_ = FilePlayer::CreateFilePlayer(_inputFilePlayerId,
- (const FileFormats)format);
-
- if (!input_file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (input_file_player_->StartPlayingFile(
- fileName, loop, startPosition, volumeScaling, notificationTime,
- stopPosition, (const CodecInst*)codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- input_file_player_->StopPlayingFile();
- input_file_player_.reset();
- return -1;
- }
- input_file_player_->RegisterModuleFileCallback(this);
- channel_state_.SetInputFilePlaying(true);
-
- return 0;
-}
-
-int Channel::StartPlayingFileAsMicrophone(InStream* stream,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartPlayingFileAsMicrophone(format=%d, "
- "volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
- format, volumeScaling, startPosition, stopPosition);
-
- if (stream == NULL) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFileAsMicrophone NULL as input stream");
- return -1;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- if (channel_state_.Get().input_file_playing) {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() is playing");
- return 0;
- }
-
- // Destroy the old instance
- if (input_file_player_) {
- input_file_player_->RegisterModuleFileCallback(NULL);
- input_file_player_.reset();
- }
-
- // Create the instance
- input_file_player_ = FilePlayer::CreateFilePlayer(_inputFilePlayerId,
- (const FileFormats)format);
-
- if (!input_file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingInputFile() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (input_file_player_->StartPlayingFile(stream, startPosition, volumeScaling,
- notificationTime, stopPosition,
- codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start "
- "file playout");
- input_file_player_->StopPlayingFile();
- input_file_player_.reset();
- return -1;
- }
-
- input_file_player_->RegisterModuleFileCallback(this);
- channel_state_.SetInputFilePlaying(true);
-
- return 0;
-}
-
-int Channel::StopPlayingFileAsMicrophone() {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StopPlayingFileAsMicrophone()");
-
- rtc::CritScope cs(&_fileCritSect);
-
- if (!channel_state_.Get().input_file_playing) {
- return 0;
- }
-
- if (input_file_player_->StopPlayingFile() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopPlayingFile() could not stop playing");
- return -1;
- }
- input_file_player_->RegisterModuleFileCallback(NULL);
- input_file_player_.reset();
- channel_state_.SetInputFilePlaying(false);
-
- return 0;
-}
-
-int Channel::IsPlayingFileAsMicrophone() const {
- return channel_state_.Get().input_file_playing;
-}
-
-int Channel::StartRecordingPlayout(const char* fileName,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartRecordingPlayout(fileName=%s)", fileName);
-
- if (_outputFileRecording) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000};
-
- if ((codecInst != NULL) &&
- ((codecInst->channels < 1) || (codecInst->channels > 2))) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return (-1);
- }
- if (codecInst == NULL) {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname, "PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname, "PCMA") == 0)) {
- format = kFileFormatWavFile;
- } else {
- format = kFileFormatCompressedFile;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- // Destroy the old instance
- if (output_file_recorder_) {
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_.reset();
- }
-
- output_file_recorder_ = FileRecorder::CreateFileRecorder(
- _outputFileRecorderId, (const FileFormats)format);
- if (!output_file_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (output_file_recorder_->StartRecordingAudioFile(
- fileName, (const CodecInst&)*codecInst, notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- output_file_recorder_->StopRecording();
- output_file_recorder_.reset();
- return -1;
- }
- output_file_recorder_->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int Channel::StartRecordingPlayout(OutStream* stream,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartRecordingPlayout()");
-
- if (_outputFileRecording) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000};
-
- if (codecInst != NULL && codecInst->channels != 1) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return (-1);
- }
- if (codecInst == NULL) {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname, "PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname, "PCMA") == 0)) {
- format = kFileFormatWavFile;
- } else {
- format = kFileFormatCompressedFile;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- // Destroy the old instance
- if (output_file_recorder_) {
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_.reset();
- }
-
- output_file_recorder_ = FileRecorder::CreateFileRecorder(
- _outputFileRecorderId, (const FileFormats)format);
- if (!output_file_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (output_file_recorder_->StartRecordingAudioFile(stream, *codecInst,
- notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartRecordingPlayout() failed to "
- "start file recording");
- output_file_recorder_->StopRecording();
- output_file_recorder_.reset();
- return -1;
- }
-
- output_file_recorder_->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int Channel::StopRecordingPlayout() {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "Channel::StopRecordingPlayout()");
-
- if (!_outputFileRecording) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StopRecordingPlayout() isnot recording");
- return -1;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- if (output_file_recorder_->StopRecording() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording() could not stop recording");
- return (-1);
- }
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_.reset();
- _outputFileRecording = false;
-
- return 0;
-}
-
-void Channel::SetMixWithMicStatus(bool mix) {
- rtc::CritScope cs(&_fileCritSect);
- _mixFileWithMicrophone = mix;
-}
-
int Channel::GetSpeechOutputLevel() const {
return _outputAudioLevel.Level();
}
@@ -2760,10 +2156,6 @@
RTC_DCHECK_LE(audio_input->num_channels_, 2);
RTC_DCHECK_EQ(audio_input->id_, ChannelId());
- if (channel_state_.Get().input_file_playing) {
- MixOrReplaceAudioWithFile(audio_input);
- }
-
bool is_muted = InputMute();
AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);
@@ -2903,96 +2295,6 @@
return 0;
}
-// TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
-// a shared helper.
-int32_t Channel::MixOrReplaceAudioWithFile(AudioFrame* audio_input) {
- RTC_DCHECK_RUN_ON(encoder_queue_);
- std::unique_ptr<int16_t[]> fileBuffer(new int16_t[640]);
- size_t fileSamples(0);
- const int mixingFrequency = audio_input->sample_rate_hz_;
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (!input_file_player_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixOrReplaceAudioWithFile() fileplayer"
- " doesnt exist");
- return -1;
- }
-
- if (input_file_player_->Get10msAudioFromFile(fileBuffer.get(), &fileSamples,
- mixingFrequency) == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixOrReplaceAudioWithFile() file mixing "
- "failed");
- return -1;
- }
- if (fileSamples == 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixOrReplaceAudioWithFile() file is ended");
- return 0;
- }
- }
-
- RTC_DCHECK_EQ(audio_input->samples_per_channel_, fileSamples);
-
- if (_mixFileWithMicrophone) {
- // Currently file stream is always mono.
- // TODO(xians): Change the code when FilePlayer supports real stereo.
- MixWithSat(audio_input->mutable_data(), audio_input->num_channels_,
- fileBuffer.get(), 1, fileSamples);
- } else {
- // Replace ACM audio with file.
- // Currently file stream is always mono.
- // TODO(xians): Change the code when FilePlayer supports real stereo.
- audio_input->UpdateFrame(
- _channelId, 0xFFFFFFFF, fileBuffer.get(), fileSamples, mixingFrequency,
- AudioFrame::kNormalSpeech, AudioFrame::kVadUnknown, 1);
- }
- return 0;
-}
-
-int32_t Channel::MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency) {
- assert(mixingFrequency <= 48000);
-
- std::unique_ptr<int16_t[]> fileBuffer(new int16_t[960]);
- size_t fileSamples(0);
-
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (!output_file_player_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixAudioWithFile() file mixing failed");
- return -1;
- }
-
- // We should get the frequency we ask for.
- if (output_file_player_->Get10msAudioFromFile(
- fileBuffer.get(), &fileSamples, mixingFrequency) == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixAudioWithFile() file mixing failed");
- return -1;
- }
- }
-
- if (audioFrame.samples_per_channel_ == fileSamples) {
- // Currently file stream is always mono.
- // TODO(xians): Change the code when FilePlayer supports real stereo.
- MixWithSat(audioFrame.mutable_data(), audioFrame.num_channels_,
- fileBuffer.get(), 1, fileSamples);
- } else {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixAudioWithFile() samples_per_channel_(%" PRIuS
- ") != "
- "fileSamples(%" PRIuS ")",
- audioFrame.samples_per_channel_, fileSamples);
- return -1;
- }
-
- return 0;
-}
-
void Channel::UpdatePlayoutTimestamp(bool rtcp) {
jitter_buffer_playout_timestamp_ = audio_coding_->PlayoutTimestamp();
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 5f2df14..d023f87 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -32,8 +32,6 @@
#include "rtc_base/event.h"
#include "rtc_base/thread_checker.h"
#include "voice_engine/audio_level.h"
-#include "voice_engine/file_player.h"
-#include "voice_engine/file_recorder.h"
#include "voice_engine/include/voe_base.h"
#include "voice_engine/include/voe_network.h"
#include "voice_engine/shared_data.h"
@@ -46,7 +44,6 @@
namespace webrtc {
class AudioDeviceModule;
-class FileWrapper;
class PacketRouter;
class ProcessThread;
class RateLimiter;
@@ -85,8 +82,6 @@
class ChannelState {
public:
struct State {
- bool output_file_playing = false;
- bool input_file_playing = false;
bool playing = false;
bool sending = false;
};
@@ -104,16 +99,6 @@
return state_;
}
- void SetOutputFilePlaying(bool enable) {
- rtc::CritScope lock(&lock_);
- state_.output_file_playing = enable;
- }
-
- void SetInputFilePlaying(bool enable) {
- rtc::CritScope lock(&lock_);
- state_.input_file_playing = enable;
- }
-
void SetPlaying(bool enable) {
rtc::CritScope lock(&lock_);
state_.playing = enable;
@@ -132,8 +117,6 @@
class Channel
: public RtpData,
public RtpFeedback,
- public FileCallback, // receiving notification from file player &
- // recorder
public Transport,
public AudioPacketizationCallback, // receive encoded packets from the
// ACM
@@ -217,44 +200,6 @@
int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
void OnRtpPacket(const RtpPacketReceived& packet);
- // VoEFile
- int StartPlayingFileLocally(const char* fileName,
- bool loop,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst);
- int StartPlayingFileLocally(InStream* stream,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst);
- int StopPlayingFileLocally();
- int IsPlayingFileLocally() const;
- int RegisterFilePlayingToMixer();
- int StartPlayingFileAsMicrophone(const char* fileName,
- bool loop,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst);
- int StartPlayingFileAsMicrophone(InStream* stream,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst);
- int StopPlayingFileAsMicrophone();
- int IsPlayingFileAsMicrophone() const;
- int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst);
- int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
- int StopRecordingPlayout();
-
- void SetMixWithMicStatus(bool mix);
-
// Muting, Volume and Level.
void SetInputMute(bool enable);
void SetChannelOutputVolumeScaling(float scaling);
@@ -348,12 +293,6 @@
int sample_rate_hz,
AudioFrame* audio_frame);
- // From FileCallback
- void PlayNotification(int32_t id, uint32_t durationMs) override;
- void RecordNotification(int32_t id, uint32_t durationMs) override;
- void PlayFileEnded(int32_t id) override;
- void RecordFileEnded(int32_t id) override;
-
uint32_t InstanceId() const { return _instanceId; }
int32_t ChannelId() const { return _channelId; }
bool Playing() const { return channel_state_.Get().playing; }
@@ -430,8 +369,6 @@
bool IsPacketInOrder(const RTPHeader& header) const;
bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
int ResendPackets(const uint16_t* sequence_numbers, int length);
- int32_t MixOrReplaceAudioWithFile(AudioFrame* audio_frame);
- int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
void UpdatePlayoutTimestamp(bool rtcp);
void RegisterReceiveCodecsToRTPModule();
@@ -452,7 +389,6 @@
uint32_t _instanceId;
int32_t _channelId;
- rtc::CriticalSection _fileCritSect;
rtc::CriticalSection _callbackCritSect;
rtc::CriticalSection volume_settings_critsect_;
@@ -475,13 +411,6 @@
bool _externalTransport;
// Downsamples to the codec rate if necessary.
PushResampler<int16_t> input_resampler_;
- std::unique_ptr<FilePlayer> input_file_player_;
- std::unique_ptr<FilePlayer> output_file_player_;
- std::unique_ptr<FileRecorder> output_file_recorder_;
- int _inputFilePlayerId;
- int _outputFilePlayerId;
- int _outputFileRecorderId;
- bool _outputFileRecording;
uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_);
RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
@@ -515,8 +444,6 @@
bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
bool previous_frame_muted_ RTC_ACCESS_ON(encoder_queue_);
float _outputGain RTC_GUARDED_BY(volume_settings_critsect_);
- // VoEBase
- bool _mixFileWithMicrophone;
// VoeRTP_RTCP
// TODO(henrika): can today be accessed on the main thread and on the
// task queue; hence potential race.
diff --git a/voice_engine/coder.cc b/voice_engine/coder.cc
deleted file mode 100644
index 9eff1d4..0000000
--- a/voice_engine/coder.cc
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voice_engine/coder.h"
-
-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "common_types.h" // NOLINT(build/include)
-#include "modules/audio_coding/codecs/audio_format_conversion.h"
-#include "modules/include/module_common_types.h"
-
-namespace webrtc {
-namespace {
-AudioCodingModule::Config GetAcmConfig(uint32_t id) {
- AudioCodingModule::Config config;
- // This class does not handle muted output.
- config.neteq_config.enable_muted_state = false;
- config.id = id;
- config.decoder_factory = CreateBuiltinAudioDecoderFactory();
- return config;
-}
-} // namespace
-
-AudioCoder::AudioCoder(uint32_t instance_id)
- : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))),
- receive_codec_(),
- encode_timestamp_(0),
- encoded_data_(nullptr),
- encoded_length_in_bytes_(0),
- decode_timestamp_(0) {
- acm_->InitializeReceiver();
- acm_->RegisterTransportCallback(this);
-}
-
-AudioCoder::~AudioCoder() {}
-
-int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) {
- const bool success = codec_manager_.RegisterEncoder(codec_inst) &&
- codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get());
- return success ? 0 : -1;
-}
-
-int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
- if (!acm_->RegisterReceiveCodec(codec_inst.pltype,
- CodecInstToSdp(codec_inst))) {
- return -1;
- }
- memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
- return 0;
-}
-
-int32_t AudioCoder::Decode(AudioFrame* decoded_audio,
- uint32_t samp_freq_hz,
- const int8_t* incoming_payload,
- size_t payload_length) {
- if (payload_length > 0) {
- const uint8_t payload_type = receive_codec_.pltype;
- decode_timestamp_ += receive_codec_.pacsize;
- if (acm_->IncomingPayload((const uint8_t*)incoming_payload, payload_length,
- payload_type, decode_timestamp_) == -1) {
- return -1;
- }
- }
- bool muted;
- int32_t ret =
- acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, decoded_audio, &muted);
- RTC_DCHECK(!muted);
- return ret;
-}
-
-int32_t AudioCoder::PlayoutData(AudioFrame* decoded_audio,
- uint16_t samp_freq_hz) {
- bool muted;
- int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted);
- RTC_DCHECK(!muted);
- return ret;
-}
-
-int32_t AudioCoder::Encode(const AudioFrame& audio,
- int8_t* encoded_data,
- size_t* encoded_length_in_bytes) {
- // Fake a timestamp in case audio doesn't contain a correct timestamp.
- // Make a local copy of the audio frame since audio is const
- AudioFrame audio_frame;
- audio_frame.CopyFrom(audio);
- audio_frame.timestamp_ = encode_timestamp_;
- encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_);
-
- // For any codec with a frame size that is longer than 10 ms the encoded
- // length in bytes should be zero until a a full frame has been encoded.
- encoded_length_in_bytes_ = 0;
- encoded_data_ = encoded_data;
- if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
- return -1;
- }
-
- *encoded_length_in_bytes = encoded_length_in_bytes_;
- return 0;
-}
-
-int32_t AudioCoder::SendData(FrameType /* frame_type */,
- uint8_t /* payload_type */,
- uint32_t /* time_stamp */,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* /* fragmentation*/) {
- memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
- encoded_length_in_bytes_ = payload_size;
- return 0;
-}
-
-} // namespace webrtc
diff --git a/voice_engine/coder.h b/voice_engine/coder.h
deleted file mode 100644
index bd65b9d..0000000
--- a/voice_engine/coder.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef VOICE_ENGINE_CODER_H_
-#define VOICE_ENGINE_CODER_H_
-
-#include <memory>
-
-#include "common_types.h" // NOLINT(build/include)
-#include "modules/audio_coding/acm2/codec_manager.h"
-#include "modules/audio_coding/acm2/rent_a_codec.h"
-#include "modules/audio_coding/include/audio_coding_module.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-class AudioFrame;
-
-class AudioCoder : public AudioPacketizationCallback {
- public:
- explicit AudioCoder(uint32_t instance_id);
- ~AudioCoder();
-
- int32_t SetEncodeCodec(const CodecInst& codec_inst);
-
- int32_t SetDecodeCodec(const CodecInst& codec_inst);
-
- int32_t Decode(AudioFrame* decoded_audio,
- uint32_t samp_freq_hz,
- const int8_t* incoming_payload,
- size_t payload_length);
-
- int32_t PlayoutData(AudioFrame* decoded_audio, uint16_t samp_freq_hz);
-
- int32_t Encode(const AudioFrame& audio,
- int8_t* encoded_data,
- size_t* encoded_length_in_bytes);
-
- protected:
- int32_t SendData(FrameType frame_type,
- uint8_t payload_type,
- uint32_t time_stamp,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation) override;
-
- private:
- std::unique_ptr<AudioCodingModule> acm_;
- acm2::CodecManager codec_manager_;
- acm2::RentACodec rent_a_codec_;
-
- CodecInst receive_codec_;
-
- uint32_t encode_timestamp_;
- int8_t* encoded_data_;
- size_t encoded_length_in_bytes_;
-
- uint32_t decode_timestamp_;
-};
-} // namespace webrtc
-
-#endif // VOICE_ENGINE_CODER_H_
diff --git a/voice_engine/file_player.cc b/voice_engine/file_player.cc
deleted file mode 100644
index 11f6ef0..0000000
--- a/voice_engine/file_player.cc
+++ /dev/null
@@ -1,397 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voice_engine/file_player.h"
-
-#include "common_audio/resampler/include/resampler.h"
-#include "common_types.h" // NOLINT(build/include)
-#include "modules/media_file/media_file.h"
-#include "modules/media_file/media_file_defines.h"
-#include "rtc_base/logging.h"
-#include "typedefs.h" // NOLINT(build/include)
-#include "voice_engine/coder.h"
-
-namespace webrtc {
-
-namespace {
-
-class FilePlayerImpl : public FilePlayer {
- public:
- FilePlayerImpl(uint32_t instanceID, FileFormats fileFormat);
- ~FilePlayerImpl() override;
-
- int Get10msAudioFromFile(int16_t* outBuffer,
- size_t* lengthInSamples,
- int frequencyInHz) override;
- int32_t RegisterModuleFileCallback(FileCallback* callback) override;
- int32_t StartPlayingFile(const char* fileName,
- bool loop,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition,
- const CodecInst* codecInst) override;
- int32_t StartPlayingFile(InStream* sourceStream,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition,
- const CodecInst* codecInst) override;
- int32_t StopPlayingFile() override;
- bool IsPlayingFile() const override;
- int32_t GetPlayoutPosition(uint32_t* durationMs) override;
- int32_t AudioCodec(CodecInst* audioCodec) const override;
- int32_t Frequency() const override;
- int32_t SetAudioScaling(float scaleFactor) override;
-
- private:
- int32_t SetUpAudioDecoder();
-
- const FileFormats _fileFormat;
- MediaFile& _fileModule;
-
- uint32_t _decodedLengthInMS;
-
- AudioCoder _audioDecoder;
-
- CodecInst _codec;
- int32_t _numberOf10MsPerFrame;
- int32_t _numberOf10MsInDecoder;
-
- Resampler _resampler;
- float _scaling;
-};
-
-FilePlayerImpl::FilePlayerImpl(const uint32_t instanceID,
- const FileFormats fileFormat)
- : _fileFormat(fileFormat),
- _fileModule(*MediaFile::CreateMediaFile(instanceID)),
- _decodedLengthInMS(0),
- _audioDecoder(instanceID),
- _codec(),
- _numberOf10MsPerFrame(0),
- _numberOf10MsInDecoder(0),
- _resampler(),
- _scaling(1.0) {
- _codec.plfreq = 0;
-}
-
-FilePlayerImpl::~FilePlayerImpl() {
- MediaFile::DestroyMediaFile(&_fileModule);
-}
-
-int32_t FilePlayerImpl::Frequency() const {
- if (_codec.plfreq == 0) {
- return -1;
- }
- // Make sure that sample rate is 8,16 or 32 kHz. E.g. WAVE files may have
- // other sampling rates.
- if (_codec.plfreq == 11000) {
- return 16000;
- } else if (_codec.plfreq == 22000) {
- return 32000;
- } else if (_codec.plfreq == 44000) {
- return 32000;
- } else if (_codec.plfreq == 48000) {
- return 48000;
- } else {
- return _codec.plfreq;
- }
-}
-
-int32_t FilePlayerImpl::AudioCodec(CodecInst* audioCodec) const {
- *audioCodec = _codec;
- return 0;
-}
-
-int32_t FilePlayerImpl::Get10msAudioFromFile(int16_t* outBuffer,
- size_t* lengthInSamples,
- int frequencyInHz) {
- if (_codec.plfreq == 0) {
- LOG(LS_WARNING) << "Get10msAudioFromFile() playing not started!"
- << " codec freq = " << _codec.plfreq
- << ", wanted freq = " << frequencyInHz;
- return -1;
- }
-
- AudioFrame unresampledAudioFrame;
- if (STR_CASE_CMP(_codec.plname, "L16") == 0) {
- unresampledAudioFrame.sample_rate_hz_ = _codec.plfreq;
-
- // L16 is un-encoded data. Just pull 10 ms.
- size_t lengthInBytes = AudioFrame::kMaxDataSizeBytes;
- if (_fileModule.PlayoutAudioData(
- reinterpret_cast<int8_t*>(unresampledAudioFrame.mutable_data()),
- lengthInBytes) == -1) {
- // End of file reached.
- return -1;
- }
- if (lengthInBytes == 0) {
- *lengthInSamples = 0;
- return 0;
- }
- // One sample is two bytes.
- unresampledAudioFrame.samples_per_channel_ = lengthInBytes >> 1;
-
- } else {
- // Decode will generate 10 ms of audio data. PlayoutAudioData(..)
- // expects a full frame. If the frame size is larger than 10 ms,
- // PlayoutAudioData(..) data should be called proportionally less often.
- int16_t encodedBuffer[MAX_AUDIO_BUFFER_IN_SAMPLES];
- size_t encodedLengthInBytes = 0;
- if (++_numberOf10MsInDecoder >= _numberOf10MsPerFrame) {
- _numberOf10MsInDecoder = 0;
- size_t bytesFromFile = sizeof(encodedBuffer);
- if (_fileModule.PlayoutAudioData(reinterpret_cast<int8_t*>(encodedBuffer),
- bytesFromFile) == -1) {
- // End of file reached.
- return -1;
- }
- encodedLengthInBytes = bytesFromFile;
- }
- if (_audioDecoder.Decode(&unresampledAudioFrame, frequencyInHz,
- reinterpret_cast<int8_t*>(encodedBuffer),
- encodedLengthInBytes) == -1) {
- return -1;
- }
- }
-
- size_t outLen = 0;
- if (_resampler.ResetIfNeeded(unresampledAudioFrame.sample_rate_hz_,
- frequencyInHz, 1)) {
- LOG(LS_WARNING) << "Get10msAudioFromFile() unexpected codec.";
-
- // New sampling frequency. Update state.
- outLen = static_cast<size_t>(frequencyInHz / 100);
- memset(outBuffer, 0, outLen * sizeof(int16_t));
- return 0;
- }
- _resampler.Push(unresampledAudioFrame.data(),
- unresampledAudioFrame.samples_per_channel_, outBuffer,
- MAX_AUDIO_BUFFER_IN_SAMPLES, outLen);
-
- *lengthInSamples = outLen;
-
- if (_scaling != 1.0) {
- for (size_t i = 0; i < outLen; i++) {
- outBuffer[i] = (int16_t)(outBuffer[i] * _scaling);
- }
- }
- _decodedLengthInMS += 10;
- return 0;
-}
-
-int32_t FilePlayerImpl::RegisterModuleFileCallback(FileCallback* callback) {
- return _fileModule.SetModuleFileCallback(callback);
-}
-
-int32_t FilePlayerImpl::SetAudioScaling(float scaleFactor) {
- if ((scaleFactor >= 0) && (scaleFactor <= 2.0)) {
- _scaling = scaleFactor;
- return 0;
- }
- LOG(LS_WARNING) << "SetAudioScaling() non-allowed scale factor.";
- return -1;
-}
-
-int32_t FilePlayerImpl::StartPlayingFile(const char* fileName,
- bool loop,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition,
- const CodecInst* codecInst) {
- if (_fileFormat == kFileFormatPcm16kHzFile ||
- _fileFormat == kFileFormatPcm8kHzFile ||
- _fileFormat == kFileFormatPcm32kHzFile) {
- CodecInst codecInstL16;
- strncpy(codecInstL16.plname, "L16", 32);
- codecInstL16.pltype = 93;
- codecInstL16.channels = 1;
-
- if (_fileFormat == kFileFormatPcm8kHzFile) {
- codecInstL16.rate = 128000;
- codecInstL16.plfreq = 8000;
- codecInstL16.pacsize = 80;
- } else if (_fileFormat == kFileFormatPcm16kHzFile) {
- codecInstL16.rate = 256000;
- codecInstL16.plfreq = 16000;
- codecInstL16.pacsize = 160;
- } else if (_fileFormat == kFileFormatPcm32kHzFile) {
- codecInstL16.rate = 512000;
- codecInstL16.plfreq = 32000;
- codecInstL16.pacsize = 320;
- } else if (_fileFormat == kFileFormatPcm48kHzFile) {
- codecInstL16.rate = 768000;
- codecInstL16.plfreq = 48000;
- codecInstL16.pacsize = 480;
- } else {
- LOG(LS_ERROR) << "StartPlayingFile() sample frequency not "
- << "supported for PCM format.";
- return -1;
- }
-
- if (_fileModule.StartPlayingAudioFile(fileName, notification, loop,
- _fileFormat, &codecInstL16,
- startPosition, stopPosition) == -1) {
- LOG(LS_WARNING) << "StartPlayingFile() failed to initialize "
- << "pcm file " << fileName;
- return -1;
- }
- SetAudioScaling(volumeScaling);
- } else if (_fileFormat == kFileFormatPreencodedFile) {
- if (_fileModule.StartPlayingAudioFile(fileName, notification, loop,
- _fileFormat, codecInst) == -1) {
- LOG(LS_WARNING) << "StartPlayingFile() failed to initialize "
- << "pre-encoded file " << fileName;
- return -1;
- }
- } else {
- CodecInst* no_inst = NULL;
- if (_fileModule.StartPlayingAudioFile(fileName, notification, loop,
- _fileFormat, no_inst, startPosition,
- stopPosition) == -1) {
- LOG(LS_WARNING) << "StartPlayingFile() failed to initialize file "
- << fileName;
- return -1;
- }
- SetAudioScaling(volumeScaling);
- }
- if (SetUpAudioDecoder() == -1) {
- StopPlayingFile();
- return -1;
- }
- return 0;
-}
-
-int32_t FilePlayerImpl::StartPlayingFile(InStream* sourceStream,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition,
- const CodecInst* codecInst) {
- if (_fileFormat == kFileFormatPcm16kHzFile ||
- _fileFormat == kFileFormatPcm32kHzFile ||
- _fileFormat == kFileFormatPcm8kHzFile ||
- _fileFormat == kFileFormatPcm48kHzFile) {
- CodecInst codecInstL16;
- strncpy(codecInstL16.plname, "L16", 32);
- codecInstL16.pltype = 93;
- codecInstL16.channels = 1;
-
- if (_fileFormat == kFileFormatPcm8kHzFile) {
- codecInstL16.rate = 128000;
- codecInstL16.plfreq = 8000;
- codecInstL16.pacsize = 80;
- } else if (_fileFormat == kFileFormatPcm16kHzFile) {
- codecInstL16.rate = 256000;
- codecInstL16.plfreq = 16000;
- codecInstL16.pacsize = 160;
- } else if (_fileFormat == kFileFormatPcm32kHzFile) {
- codecInstL16.rate = 512000;
- codecInstL16.plfreq = 32000;
- codecInstL16.pacsize = 320;
- } else if (_fileFormat == kFileFormatPcm48kHzFile) {
- codecInstL16.rate = 768000;
- codecInstL16.plfreq = 48000;
- codecInstL16.pacsize = 480;
- } else {
- LOG(LS_ERROR) << "StartPlayingFile() sample frequency not "
- << "supported for PCM format.";
- return -1;
- }
- if (_fileModule.StartPlayingAudioStream(
- *sourceStream, notification, _fileFormat, &codecInstL16,
- startPosition, stopPosition) == -1) {
- LOG(LS_ERROR) << "StartPlayingFile() failed to initialize stream "
- << "playout.";
- return -1;
- }
-
- } else if (_fileFormat == kFileFormatPreencodedFile) {
- if (_fileModule.StartPlayingAudioStream(*sourceStream, notification,
- _fileFormat, codecInst) == -1) {
- LOG(LS_ERROR) << "StartPlayingFile() failed to initialize stream "
- << "playout.";
- return -1;
- }
- } else {
- CodecInst* no_inst = NULL;
- if (_fileModule.StartPlayingAudioStream(*sourceStream, notification,
- _fileFormat, no_inst, startPosition,
- stopPosition) == -1) {
- LOG(LS_ERROR) << "StartPlayingFile() failed to initialize stream "
- << "playout.";
- return -1;
- }
- }
- SetAudioScaling(volumeScaling);
-
- if (SetUpAudioDecoder() == -1) {
- StopPlayingFile();
- return -1;
- }
- return 0;
-}
-
-int32_t FilePlayerImpl::StopPlayingFile() {
- memset(&_codec, 0, sizeof(CodecInst));
- _numberOf10MsPerFrame = 0;
- _numberOf10MsInDecoder = 0;
- return _fileModule.StopPlaying();
-}
-
-bool FilePlayerImpl::IsPlayingFile() const {
- return _fileModule.IsPlaying();
-}
-
-int32_t FilePlayerImpl::GetPlayoutPosition(uint32_t* durationMs) {
- return _fileModule.PlayoutPositionMs(*durationMs);
-}
-
-int32_t FilePlayerImpl::SetUpAudioDecoder() {
- if ((_fileModule.codec_info(_codec) == -1)) {
- LOG(LS_WARNING) << "Failed to retrieve codec info of file data.";
- return -1;
- }
- if (STR_CASE_CMP(_codec.plname, "L16") != 0 &&
- _audioDecoder.SetDecodeCodec(_codec) == -1) {
- LOG(LS_WARNING) << "SetUpAudioDecoder() codec " << _codec.plname
- << " not supported.";
- return -1;
- }
- _numberOf10MsPerFrame = _codec.pacsize / (_codec.plfreq / 100);
- _numberOf10MsInDecoder = 0;
- return 0;
-}
-
-} // namespace
-
-std::unique_ptr<FilePlayer> FilePlayer::CreateFilePlayer(
- uint32_t instanceID,
- FileFormats fileFormat) {
- switch (fileFormat) {
- case kFileFormatWavFile:
- case kFileFormatCompressedFile:
- case kFileFormatPreencodedFile:
- case kFileFormatPcm16kHzFile:
- case kFileFormatPcm8kHzFile:
- case kFileFormatPcm32kHzFile:
- case kFileFormatPcm48kHzFile:
- // audio formats
- return std::unique_ptr<FilePlayer>(
- new FilePlayerImpl(instanceID, fileFormat));
- default:
- assert(false);
- return nullptr;
- }
-}
-
-} // namespace webrtc
diff --git a/voice_engine/file_player.h b/voice_engine/file_player.h
deleted file mode 100644
index 252e1b3..0000000
--- a/voice_engine/file_player.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef VOICE_ENGINE_FILE_PLAYER_H_
-#define VOICE_ENGINE_FILE_PLAYER_H_
-
-#include <memory>
-
-#include "common_types.h" // NOLINT(build/include)
-#include "modules/include/module_common_types.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-
-class FileCallback;
-
-class FilePlayer {
- public:
- // The largest decoded frame size in samples (60ms with 48kHz sample rate).
- enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 48 };
- enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
-
- // Note: will return NULL for unsupported formats.
- static std::unique_ptr<FilePlayer> CreateFilePlayer(
- const uint32_t instanceID,
- const FileFormats fileFormat);
-
- virtual ~FilePlayer() = default;
-
- // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
- // will be set to the number of samples read (not the number of samples per
- // channel).
- virtual int Get10msAudioFromFile(int16_t* outBuffer,
- size_t* lengthInSamples,
- int frequencyInHz) = 0;
-
- // Register callback for receiving file playing notifications.
- virtual int32_t RegisterModuleFileCallback(FileCallback* callback) = 0;
-
- // API for playing audio from fileName to channel.
- // Note: codecInst is used for pre-encoded files.
- virtual int32_t StartPlayingFile(const char* fileName,
- bool loop,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition,
- const CodecInst* codecInst) = 0;
-
- // Note: codecInst is used for pre-encoded files.
- virtual int32_t StartPlayingFile(InStream* sourceStream,
- uint32_t startPosition,
- float volumeScaling,
- uint32_t notification,
- uint32_t stopPosition,
- const CodecInst* codecInst) = 0;
-
- virtual int32_t StopPlayingFile() = 0;
-
- virtual bool IsPlayingFile() const = 0;
-
- virtual int32_t GetPlayoutPosition(uint32_t* durationMs) = 0;
-
- // Set audioCodec to the currently used audio codec.
- virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0;
-
- virtual int32_t Frequency() const = 0;
-
- // Note: scaleFactor is in the range [0.0 - 2.0]
- virtual int32_t SetAudioScaling(float scaleFactor) = 0;
-};
-} // namespace webrtc
-#endif // VOICE_ENGINE_FILE_PLAYER_H_
diff --git a/voice_engine/file_player_unittests.cc b/voice_engine/file_player_unittests.cc
deleted file mode 100644
index 27a3eae..0000000
--- a/voice_engine/file_player_unittests.cc
+++ /dev/null
@@ -1,115 +0,0 @@
-/*
- * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Unit tests for FilePlayer.
-
-#include "voice_engine/file_player.h"
-
-#include <stdio.h>
-
-#include <memory>
-#include <string>
-
-#include "rtc_base/flags.h"
-#include "rtc_base/md5digest.h"
-#include "rtc_base/stringencode.h"
-#include "test/gtest.h"
-#include "test/testsupport/fileutils.h"
-
-DEFINE_bool(file_player_output, false, "Generate reference files.");
-
-namespace webrtc {
-
-class FilePlayerTest : public ::testing::Test {
- protected:
- static const uint32_t kId = 0;
- static const FileFormats kFileFormat = kFileFormatWavFile;
- static const int kSampleRateHz = 8000;
-
- FilePlayerTest()
- : player_(FilePlayer::CreateFilePlayer(kId, kFileFormat)),
- output_file_(NULL) {}
-
- void SetUp() override {
- if (FLAG_file_player_output) {
- std::string output_file =
- webrtc::test::OutputPath() + "file_player_unittest_out.pcm";
- output_file_ = fopen(output_file.c_str(), "wb");
- ASSERT_TRUE(output_file_ != NULL);
- }
- }
-
- void TearDown() override {
- if (output_file_)
- fclose(output_file_);
- }
-
- void PlayFileAndCheck(const std::string& input_file,
- const std::string& ref_checksum,
- int output_length_ms) {
- const float kScaling = 1;
- ASSERT_EQ(0, player_->StartPlayingFile(input_file.c_str(), false, 0,
- kScaling, 0, 0, NULL));
- rtc::Md5Digest checksum;
- for (int i = 0; i < output_length_ms / 10; ++i) {
- int16_t out[10 * kSampleRateHz / 1000] = {0};
- size_t num_samples;
- EXPECT_EQ(
- 0, player_->Get10msAudioFromFile(out, &num_samples, kSampleRateHz));
- checksum.Update(out, num_samples * sizeof(out[0]));
- if (FLAG_file_player_output) {
- ASSERT_EQ(num_samples,
- fwrite(out, sizeof(out[0]), num_samples, output_file_));
- }
- }
- char checksum_result[rtc::Md5Digest::kSize];
- EXPECT_EQ(rtc::Md5Digest::kSize,
- checksum.Finish(checksum_result, rtc::Md5Digest::kSize));
- EXPECT_EQ(ref_checksum,
- rtc::hex_encode(checksum_result, sizeof(checksum_result)));
- }
-
- std::unique_ptr<FilePlayer> player_;
- FILE* output_file_;
-};
-
-#if defined(WEBRTC_IOS)
-#define MAYBE_PlayWavPcmuFile DISABLED_PlayWavPcmuFile
-#else
-#define MAYBE_PlayWavPcmuFile PlayWavPcmuFile
-#endif
-TEST_F(FilePlayerTest, MAYBE_PlayWavPcmuFile) {
- const std::string kFileName =
- test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav");
- // The file is longer than this, but keeping the output shorter limits the
- // runtime for the test.
- const int kOutputLengthMs = 10000;
- const std::string kRefChecksum = "c74e7fd432d439b1311e1c16815b3e9a";
-
- PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
-}
-
-#if defined(WEBRTC_IOS)
-#define MAYBE_PlayWavPcm16File DISABLED_PlayWavPcm16File
-#else
-#define MAYBE_PlayWavPcm16File PlayWavPcm16File
-#endif
-TEST_F(FilePlayerTest, MAYBE_PlayWavPcm16File) {
- const std::string kFileName =
- test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav");
- // The file is longer than this, but keeping the output shorter limits the
- // runtime for the test.
- const int kOutputLengthMs = 10000;
- const std::string kRefChecksum = "e41d7e1dac8aeae9f21e8e03cd7ecd71";
-
- PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
-}
-
-} // namespace webrtc
diff --git a/voice_engine/file_recorder.cc b/voice_engine/file_recorder.cc
deleted file mode 100644
index 025f52c..0000000
--- a/voice_engine/file_recorder.cc
+++ /dev/null
@@ -1,260 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voice_engine/file_recorder.h"
-
-#include <list>
-
-#include "audio/utility/audio_frame_operations.h"
-#include "common_audio/resampler/include/resampler.h"
-#include "common_types.h" // NOLINT(build/include)
-#include "modules/include/module_common_types.h"
-#include "modules/media_file/media_file.h"
-#include "modules/media_file/media_file_defines.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/platform_thread.h"
-#include "system_wrappers/include/event_wrapper.h"
-#include "typedefs.h" // NOLINT(build/include)
-#include "voice_engine/coder.h"
-
-namespace webrtc {
-
-namespace {
-
-// The largest decoded frame size in samples (60ms with 32kHz sample rate).
-enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 };
-enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
-enum { kMaxAudioBufferQueueLength = 100 };
-
-class FileRecorderImpl : public FileRecorder {
- public:
- FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat);
- ~FileRecorderImpl() override;
-
- // FileRecorder functions.
- int32_t RegisterModuleFileCallback(FileCallback* callback) override;
- FileFormats RecordingFileFormat() const override;
- int32_t StartRecordingAudioFile(const char* fileName,
- const CodecInst& codecInst,
- uint32_t notificationTimeMs) override;
- int32_t StartRecordingAudioFile(OutStream* destStream,
- const CodecInst& codecInst,
- uint32_t notificationTimeMs) override;
- int32_t StopRecording() override;
- bool IsRecording() const override;
- int32_t codec_info(CodecInst* codecInst) const override;
- int32_t RecordAudioToFile(const AudioFrame& frame) override;
-
- private:
- int32_t WriteEncodedAudioData(const int8_t* audioBuffer, size_t bufferLength);
-
- int32_t SetUpAudioEncoder();
-
- uint32_t _instanceID;
- FileFormats _fileFormat;
- MediaFile* _moduleFile;
-
- CodecInst codec_info_;
- int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
- AudioCoder _audioEncoder;
- Resampler _audioResampler;
-};
-
-FileRecorderImpl::FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat)
- : _instanceID(instanceID),
- _fileFormat(fileFormat),
- _moduleFile(MediaFile::CreateMediaFile(_instanceID)),
- codec_info_(),
- _audioBuffer(),
- _audioEncoder(instanceID),
- _audioResampler() {}
-
-FileRecorderImpl::~FileRecorderImpl() {
- MediaFile::DestroyMediaFile(_moduleFile);
-}
-
-FileFormats FileRecorderImpl::RecordingFileFormat() const {
- return _fileFormat;
-}
-
-int32_t FileRecorderImpl::RegisterModuleFileCallback(FileCallback* callback) {
- if (_moduleFile == NULL) {
- return -1;
- }
- return _moduleFile->SetModuleFileCallback(callback);
-}
-
-int32_t FileRecorderImpl::StartRecordingAudioFile(const char* fileName,
- const CodecInst& codecInst,
- uint32_t notificationTimeMs) {
- if (_moduleFile == NULL) {
- return -1;
- }
- codec_info_ = codecInst;
- int32_t retVal = 0;
- retVal = _moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
- codecInst, notificationTimeMs);
-
- if (retVal == 0) {
- retVal = SetUpAudioEncoder();
- }
- if (retVal != 0) {
- LOG(LS_WARNING) << "Failed to initialize file " << fileName
- << " for recording.";
-
- if (IsRecording()) {
- StopRecording();
- }
- }
- return retVal;
-}
-
-int32_t FileRecorderImpl::StartRecordingAudioFile(OutStream* destStream,
- const CodecInst& codecInst,
- uint32_t notificationTimeMs) {
- codec_info_ = codecInst;
- int32_t retVal = _moduleFile->StartRecordingAudioStream(
- *destStream, _fileFormat, codecInst, notificationTimeMs);
-
- if (retVal == 0) {
- retVal = SetUpAudioEncoder();
- }
- if (retVal != 0) {
- LOG(LS_WARNING) << "Failed to initialize outStream for recording.";
-
- if (IsRecording()) {
- StopRecording();
- }
- }
- return retVal;
-}
-
-int32_t FileRecorderImpl::StopRecording() {
- memset(&codec_info_, 0, sizeof(CodecInst));
- return _moduleFile->StopRecording();
-}
-
-bool FileRecorderImpl::IsRecording() const {
- return _moduleFile->IsRecording();
-}
-
-int32_t FileRecorderImpl::RecordAudioToFile(
- const AudioFrame& incomingAudioFrame) {
- if (codec_info_.plfreq == 0) {
- LOG(LS_WARNING) << "RecordAudioToFile() recording audio is not "
- << "turned on.";
- return -1;
- }
- AudioFrame tempAudioFrame;
- tempAudioFrame.samples_per_channel_ = 0;
- if (incomingAudioFrame.num_channels_ == 2 && !_moduleFile->IsStereo()) {
- // Recording mono but incoming audio is (interleaved) stereo.
- tempAudioFrame.num_channels_ = 1;
- tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
- tempAudioFrame.samples_per_channel_ =
- incomingAudioFrame.samples_per_channel_;
- if (!incomingAudioFrame.muted()) {
- AudioFrameOperations::StereoToMono(
- incomingAudioFrame.data(), incomingAudioFrame.samples_per_channel_,
- tempAudioFrame.mutable_data());
- }
- } else if (incomingAudioFrame.num_channels_ == 1 && _moduleFile->IsStereo()) {
- // Recording stereo but incoming audio is mono.
- tempAudioFrame.num_channels_ = 2;
- tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
- tempAudioFrame.samples_per_channel_ =
- incomingAudioFrame.samples_per_channel_;
- if (!incomingAudioFrame.muted()) {
- AudioFrameOperations::MonoToStereo(
- incomingAudioFrame.data(), incomingAudioFrame.samples_per_channel_,
- tempAudioFrame.mutable_data());
- }
- }
-
- const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
- if (tempAudioFrame.samples_per_channel_ != 0) {
- // If ptrAudioFrame is not empty it contains the audio to be recorded.
- ptrAudioFrame = &tempAudioFrame;
- }
-
- // Encode the audio data before writing to file. Don't encode if the codec
- // is PCM.
- // NOTE: stereo recording is only supported for WAV files.
- // TODO(hellner): WAV expect PCM in little endian byte order. Not
- // "encoding" with PCM coder should be a problem for big endian systems.
- size_t encodedLenInBytes = 0;
- if (_fileFormat == kFileFormatPreencodedFile ||
- STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
- if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
- &encodedLenInBytes) == -1) {
- LOG(LS_WARNING) << "RecordAudioToFile() codec " << codec_info_.plname
- << " not supported or failed to encode stream.";
- return -1;
- }
- } else {
- size_t outLen = 0;
- _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
- codec_info_.plfreq,
- ptrAudioFrame->num_channels_);
- // TODO(yujo): skip resample if frame is muted.
- _audioResampler.Push(
- ptrAudioFrame->data(),
- ptrAudioFrame->samples_per_channel_ * ptrAudioFrame->num_channels_,
- reinterpret_cast<int16_t*>(_audioBuffer), MAX_AUDIO_BUFFER_IN_BYTES,
- outLen);
- encodedLenInBytes = outLen * sizeof(int16_t);
- }
-
- // Codec may not be operating at a frame rate of 10 ms. Whenever enough
- // 10 ms chunks of data has been pushed to the encoder an encoded frame
- // will be available. Wait until then.
- if (encodedLenInBytes) {
- if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes) == -1) {
- return -1;
- }
- }
- return 0;
-}
-
-int32_t FileRecorderImpl::SetUpAudioEncoder() {
- if (_fileFormat == kFileFormatPreencodedFile ||
- STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
- if (_audioEncoder.SetEncodeCodec(codec_info_) == -1) {
- LOG(LS_ERROR) << "SetUpAudioEncoder() codec " << codec_info_.plname
- << " not supported.";
- return -1;
- }
- }
- return 0;
-}
-
-int32_t FileRecorderImpl::codec_info(CodecInst* codecInst) const {
- if (codec_info_.plfreq == 0) {
- return -1;
- }
- *codecInst = codec_info_;
- return 0;
-}
-
-int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
- size_t bufferLength) {
- return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
-}
-
-} // namespace
-
-std::unique_ptr<FileRecorder> FileRecorder::CreateFileRecorder(
- uint32_t instanceID,
- FileFormats fileFormat) {
- return std::unique_ptr<FileRecorder>(
- new FileRecorderImpl(instanceID, fileFormat));
-}
-
-} // namespace webrtc
diff --git a/voice_engine/file_recorder.h b/voice_engine/file_recorder.h
deleted file mode 100644
index 9c17caf..0000000
--- a/voice_engine/file_recorder.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef VOICE_ENGINE_FILE_RECORDER_H_
-#define VOICE_ENGINE_FILE_RECORDER_H_
-
-#include <memory>
-
-#include "common_types.h" // NOLINT(build/include)
-#include "modules/include/module_common_types.h"
-#include "modules/media_file/media_file_defines.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-
-class FileRecorder {
- public:
- // Note: will return NULL for unsupported formats.
- static std::unique_ptr<FileRecorder> CreateFileRecorder(
- const uint32_t instanceID,
- const FileFormats fileFormat);
-
- virtual ~FileRecorder() = default;
-
- virtual int32_t RegisterModuleFileCallback(FileCallback* callback) = 0;
-
- virtual FileFormats RecordingFileFormat() const = 0;
-
- virtual int32_t StartRecordingAudioFile(const char* fileName,
- const CodecInst& codecInst,
- uint32_t notification) = 0;
-
- virtual int32_t StartRecordingAudioFile(OutStream* destStream,
- const CodecInst& codecInst,
- uint32_t notification) = 0;
-
- // Stop recording.
- virtual int32_t StopRecording() = 0;
-
- // Return true if recording.
- virtual bool IsRecording() const = 0;
-
- virtual int32_t codec_info(CodecInst* codecInst) const = 0;
-
- // Write frame to file. Frame should contain 10ms of un-ecoded audio data.
- virtual int32_t RecordAudioToFile(const AudioFrame& frame) = 0;
-};
-
-} // namespace webrtc
-#endif // VOICE_ENGINE_FILE_RECORDER_H_
diff --git a/voice_engine/include/voe_file.h b/voice_engine/include/voe_file.h
deleted file mode 100644
index fe39f82..0000000
--- a/voice_engine/include/voe_file.h
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - File playback.
-// - File recording.
-// - File conversion.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoEFile* file = VoEFile::GetInterface(voe);
-// base->Init();
-// int ch = base->CreateChannel();
-// ...
-// base->StartPlayout(ch);
-// file->StartPlayingFileAsMicrophone(ch, "data_file_16kHz.pcm", true);
-// ...
-// file->StopPlayingFileAsMicrophone(ch);
-// base->StopPlayout(ch);
-// ...
-// base->DeleteChannel(ch);
-// base->Terminate();
-// base->Release();
-// file->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef VOICE_ENGINE_VOE_FILE_H_
-#define VOICE_ENGINE_VOE_FILE_H_
-
-#include "common_types.h" // NOLINT(build/include)
-
-namespace webrtc {
-
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoEFile {
- public:
- // Factory for the VoEFile sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEFile* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEFile sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Starts playing and mixing files with the local speaker signal for
- // playout.
- virtual int StartPlayingFileLocally(
- int channel,
- const char fileNameUTF8[1024],
- bool loop = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0,
- int stopPointMs = 0) = 0;
-
- // Starts playing and mixing streams with the local speaker signal for
- // playout.
- virtual int StartPlayingFileLocally(
- int channel,
- InStream* stream,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0,
- int stopPointMs = 0) = 0;
-
- // Stops playback of a file on a specific |channel|.
- virtual int StopPlayingFileLocally(int channel) = 0;
-
- // Returns the current file playing state for a specific |channel|.
- virtual int IsPlayingFileLocally(int channel) = 0;
-
- // Starts reading data from a file and transmits the data either
- // mixed with or instead of the microphone signal.
- virtual int StartPlayingFileAsMicrophone(
- int channel,
- const char fileNameUTF8[1024],
- bool loop = false,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0) = 0;
-
- // Starts reading data from a stream and transmits the data either
- // mixed with or instead of the microphone signal.
- virtual int StartPlayingFileAsMicrophone(
- int channel,
- InStream* stream,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0) = 0;
-
- // Stops playing of a file as microphone signal for a specific |channel|.
- virtual int StopPlayingFileAsMicrophone(int channel) = 0;
-
- // Returns whether the |channel| is currently playing a file as microphone.
- virtual int IsPlayingFileAsMicrophone(int channel) = 0;
-
- // Starts recording the mixed playout audio.
- virtual int StartRecordingPlayout(int channel,
- const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1) = 0;
-
- // Stops recording the mixed playout audio.
- virtual int StopRecordingPlayout(int channel) = 0;
-
- virtual int StartRecordingPlayout(int channel,
- OutStream* stream,
- CodecInst* compression = NULL) = 0;
-
- // Starts recording the microphone signal to a file.
- virtual int StartRecordingMicrophone(const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1) = 0;
-
- // Starts recording the microphone signal to a stream.
- virtual int StartRecordingMicrophone(OutStream* stream,
- CodecInst* compression = NULL) = 0;
-
- // Stops recording the microphone signal.
- virtual int StopRecordingMicrophone() = 0;
-
- protected:
- VoEFile() {}
- virtual ~VoEFile() {}
-};
-
-} // namespace webrtc
-
-#endif // VOICE_ENGINE_VOE_FILE_H_
diff --git a/voice_engine/output_mixer.cc b/voice_engine/output_mixer.cc
index b90dc3e..8d1e31d 100644
--- a/voice_engine/output_mixer.cc
+++ b/voice_engine/output_mixer.cc
@@ -12,7 +12,6 @@
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/format_macros.h"
-#include "system_wrappers/include/file_wrapper.h"
#include "system_wrappers/include/trace.h"
#include "voice_engine/statistics.h"
#include "voice_engine/utility.h"
@@ -33,45 +32,6 @@
_audioFrame.id_ = id;
}
-void OutputMixer::PlayNotification(int32_t id, uint32_t durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::PlayNotification(id=%d, durationMs=%d)",
- id, durationMs);
- // Not implement yet
-}
-
-void OutputMixer::RecordNotification(int32_t id,
- uint32_t durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::RecordNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void OutputMixer::PlayFileEnded(int32_t id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::PlayFileEnded(id=%d)", id);
-
- // not needed
-}
-
-void OutputMixer::RecordFileEnded(int32_t id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::RecordFileEnded(id=%d)", id);
- assert(id == _instanceId);
-
- rtc::CritScope cs(&_fileCritSect);
- _outputFileRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::RecordFileEnded() =>"
- "output file recorder module is shutdown");
-}
-
int32_t
OutputMixer::Create(OutputMixer*& mixer, uint32_t instanceId)
{
@@ -91,8 +51,7 @@
OutputMixer::OutputMixer(uint32_t instanceId) :
_mixerModule(*AudioConferenceMixer::Create(instanceId)),
_instanceId(instanceId),
- _mixingFrequencyHz(8000),
- _outputFileRecording(false)
+ _mixingFrequencyHz(8000)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::OutputMixer() - ctor");
@@ -119,13 +78,6 @@
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::~OutputMixer() - dtor");
- {
- rtc::CritScope cs(&_fileCritSect);
- if (output_file_recorder_) {
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_->StopRecording();
- }
- }
_mixerModule.UnRegisterMixedStreamCallback();
delete &_mixerModule;
}
@@ -157,192 +109,12 @@
}
int32_t
-OutputMixer::SetAnonymousMixabilityStatus(MixerParticipant& participant,
- bool mixable)
-{
- return _mixerModule.SetAnonymousMixabilityStatus(&participant, mixable);
-}
-
-int32_t
OutputMixer::MixActiveChannels()
{
_mixerModule.Process();
return 0;
}
-int OutputMixer::StartRecordingPlayout(const char* fileName,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::StartRecordingPlayout(fileName=%s)", fileName);
-
- if (_outputFileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0);
- CodecInst dummyCodec={100,"L16",16000,320,1,320000};
-
- if ((codecInst != NULL) &&
- ((codecInst->channels < 1) || (codecInst->channels > 2)))
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return(-1);
- }
- if(codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst=&dummyCodec;
- }
- else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- }
- else
- {
- format = kFileFormatCompressedFile;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- // Destroy the old instance
- if (output_file_recorder_) {
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_.reset();
- }
-
- output_file_recorder_ = FileRecorder::CreateFileRecorder(
- _instanceId, (const FileFormats)format);
- if (!output_file_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (output_file_recorder_->StartRecordingAudioFile(
- fileName, (const CodecInst&)*codecInst, notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- output_file_recorder_->StopRecording();
- output_file_recorder_.reset();
- return -1;
- }
- output_file_recorder_->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int OutputMixer::StartRecordingPlayout(OutStream* stream,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::StartRecordingPlayout()");
-
- if (_outputFileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0);
- CodecInst dummyCodec={100,"L16",16000,320,1,320000};
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return(-1);
- }
- if(codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst=&dummyCodec;
- }
- else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- }
- else
- {
- format = kFileFormatCompressedFile;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- // Destroy the old instance
- if (output_file_recorder_) {
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_.reset();
- }
-
- output_file_recorder_ = FileRecorder::CreateFileRecorder(
- _instanceId, (const FileFormats)format);
- if (!output_file_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (output_file_recorder_->StartRecordingAudioFile(stream, *codecInst,
- notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- output_file_recorder_->StopRecording();
- output_file_recorder_.reset();
- return -1;
- }
-
- output_file_recorder_->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int OutputMixer::StopRecordingPlayout()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::StopRecordingPlayout()");
-
- if (!_outputFileRecording)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StopRecordingPlayout() file isnot recording");
- return -1;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- if (output_file_recorder_->StopRecording() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording(), could not stop recording");
- return -1;
- }
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_.reset();
- _outputFileRecording = false;
-
- return 0;
-}
-
int OutputMixer::GetMixedAudio(int sample_rate_hz,
size_t num_channels,
AudioFrame* frame) {
@@ -351,13 +123,6 @@
"OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%" PRIuS ")",
sample_rate_hz, num_channels);
- // --- Record playout if enabled
- {
- rtc::CritScope cs(&_fileCritSect);
- if (_outputFileRecording && output_file_recorder_)
- output_file_recorder_->RecordAudioToFile(_audioFrame);
- }
-
frame->num_channels_ = num_channels;
frame->sample_rate_hz_ = sample_rate_hz;
// TODO(andrew): Ideally the downmixing would occur much earlier, in
diff --git a/voice_engine/output_mixer.h b/voice_engine/output_mixer.h
index b34b663..284f92d 100644
--- a/voice_engine/output_mixer.h
+++ b/voice_engine/output_mixer.h
@@ -18,7 +18,6 @@
#include "modules/audio_conference_mixer/include/audio_conference_mixer.h"
#include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "rtc_base/criticalsection.h"
-#include "voice_engine/file_recorder.h"
namespace webrtc {
@@ -29,8 +28,7 @@
class Statistics;
-class OutputMixer : public AudioMixerOutputReceiver,
- public FileCallback
+class OutputMixer : public AudioMixerOutputReceiver
{
public:
static int32_t Create(OutputMixer*& mixer, uint32_t instanceId);
@@ -49,20 +47,9 @@
int32_t SetMixabilityStatus(MixerParticipant& participant,
bool mixable);
- int32_t SetAnonymousMixabilityStatus(MixerParticipant& participant,
- bool mixable);
-
int GetMixedAudio(int sample_rate_hz, size_t num_channels,
AudioFrame* audioFrame);
- // VoEFile
- int StartRecordingPlayout(const char* fileName,
- const CodecInst* codecInst);
-
- int StartRecordingPlayout(OutStream* stream,
- const CodecInst* codecInst);
- int StopRecordingPlayout();
-
virtual ~OutputMixer();
// from AudioMixerOutputReceiver
@@ -72,14 +59,6 @@
const AudioFrame** uniqueAudioFrames,
uint32_t size);
- // For file recording
- void PlayNotification(int32_t id, uint32_t durationMs);
-
- void RecordNotification(int32_t id, uint32_t durationMs);
-
- void PlayFileEnded(int32_t id);
- void RecordFileEnded(int32_t id);
-
private:
OutputMixer(uint32_t instanceId);
@@ -87,8 +66,6 @@
Statistics* _engineStatisticsPtr;
AudioProcessing* _audioProcessingModulePtr;
- // Protects output_file_recorder_ and _outputFileRecording.
- rtc::CriticalSection _fileCritSect;
AudioConferenceMixer& _mixerModule;
AudioFrame _audioFrame;
// Converts mixed audio to the audio device output rate.
@@ -97,8 +74,6 @@
PushResampler<int16_t> audioproc_resampler_;
int _instanceId;
int _mixingFrequencyHz;
- std::unique_ptr<FileRecorder> output_file_recorder_;
- bool _outputFileRecording;
};
} // namespace voe
diff --git a/voice_engine/transmit_mixer.cc b/voice_engine/transmit_mixer.cc
index fefbc47..a888eb1 100644
--- a/voice_engine/transmit_mixer.cc
+++ b/voice_engine/transmit_mixer.cc
@@ -67,64 +67,6 @@
}
#endif // WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-void TransmitMixer::PlayNotification(int32_t id,
- uint32_t durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PlayNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void TransmitMixer::RecordNotification(int32_t id,
- uint32_t durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "TransmitMixer::RecordNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void TransmitMixer::PlayFileEnded(int32_t id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PlayFileEnded(id=%d)", id);
-
- assert(id == _filePlayerId);
-
- rtc::CritScope cs(&_critSect);
-
- _filePlaying = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PlayFileEnded() =>"
- "file player module is shutdown");
-}
-
-void
-TransmitMixer::RecordFileEnded(int32_t id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordFileEnded(id=%d)", id);
-
- if (id == _fileRecorderId)
- {
- rtc::CritScope cs(&_critSect);
- _fileRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordFileEnded() => fileRecorder module"
- "is shutdown");
- } else if (id == _fileCallRecorderId)
- {
- rtc::CritScope cs(&_critSect);
- _fileCallRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordFileEnded() => fileCallRecorder"
- "module is shutdown");
- }
-}
-
int32_t
TransmitMixer::Create(TransmitMixer*& mixer, uint32_t instanceId)
{
@@ -152,11 +94,6 @@
}
TransmitMixer::TransmitMixer(uint32_t instanceId) :
- // Avoid conflict with other channels by adding 1024 - 1026,
- // won't use as much as 1024 channels.
- _filePlayerId(instanceId + 1024),
- _fileRecorderId(instanceId + 1025),
- _fileCallRecorderId(instanceId + 1026),
#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
_monitorModule(this),
#endif
@@ -174,21 +111,6 @@
if (_processThreadPtr)
_processThreadPtr->DeRegisterModule(&_monitorModule);
#endif
- {
- rtc::CritScope cs(&_critSect);
- if (file_recorder_) {
- file_recorder_->RegisterModuleFileCallback(NULL);
- file_recorder_->StopRecording();
- }
- if (file_call_recorder_) {
- file_call_recorder_->RegisterModuleFileCallback(NULL);
- file_call_recorder_->StopRecording();
- }
- if (file_player_) {
- file_player_->RegisterModuleFileCallback(NULL);
- file_player_->StopPlayingFile();
- }
- }
}
int32_t
@@ -295,23 +217,6 @@
TypingDetection(keyPressed);
#endif
- // --- Mix with file (does not affect the mixing frequency)
- if (_filePlaying)
- {
- MixOrReplaceAudioWithFile(_audioFrame.sample_rate_hz_);
- }
-
- // --- Record to file
- bool file_recording = false;
- {
- rtc::CritScope cs(&_critSect);
- file_recording = _fileRecording;
- }
- if (file_recording)
- {
- RecordAudioToFile(_audioFrame.sample_rate_hz_);
- }
-
// --- Measure audio level of speech after all processing.
double sample_duration = static_cast<double>(nSamples) / samplesPerSec;
_audioLevel.ComputeLevel(_audioFrame, sample_duration);
@@ -344,509 +249,6 @@
return 0;
}
-int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName,
- bool loop,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartPlayingFileAsMicrophone("
- "fileNameUTF8[]=%s,loop=%d, format=%d, volumeScaling=%5.3f,"
- " startPosition=%d, stopPosition=%d)", fileName, loop,
- format, volumeScaling, startPosition, stopPosition);
-
- if (_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() is already playing");
- return 0;
- }
-
- rtc::CritScope cs(&_critSect);
-
- // Destroy the old instance
- if (file_player_) {
- file_player_->RegisterModuleFileCallback(NULL);
- file_player_.reset();
- }
-
- // Dynamically create the instance
- file_player_ =
- FilePlayer::CreateFilePlayer(_filePlayerId, (const FileFormats)format);
-
- if (!file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (file_player_->StartPlayingFile(
- fileName, loop, startPosition, volumeScaling, notificationTime,
- stopPosition, (const CodecInst*)codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- file_player_->StopPlayingFile();
- file_player_.reset();
- return -1;
- }
-
- file_player_->RegisterModuleFileCallback(this);
- _filePlaying = true;
-
- return 0;
-}
-
-int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "TransmitMixer::StartPlayingFileAsMicrophone(format=%d,"
- " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
- format, volumeScaling, startPosition, stopPosition);
-
- if (stream == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFileAsMicrophone() NULL as input stream");
- return -1;
- }
-
- if (_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() is already playing");
- return 0;
- }
-
- rtc::CritScope cs(&_critSect);
-
- // Destroy the old instance
- if (file_player_) {
- file_player_->RegisterModuleFileCallback(NULL);
- file_player_.reset();
- }
-
- // Dynamically create the instance
- file_player_ =
- FilePlayer::CreateFilePlayer(_filePlayerId, (const FileFormats)format);
-
- if (!file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceWarning,
- "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (file_player_->StartPlayingFile(stream, startPosition, volumeScaling,
- notificationTime, stopPosition,
- (const CodecInst*)codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- file_player_->StopPlayingFile();
- file_player_.reset();
- return -1;
- }
- file_player_->RegisterModuleFileCallback(this);
- _filePlaying = true;
-
- return 0;
-}
-
-int TransmitMixer::StopPlayingFileAsMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "TransmitMixer::StopPlayingFileAsMicrophone()");
-
- if (!_filePlaying)
- {
- return 0;
- }
-
- rtc::CritScope cs(&_critSect);
-
- if (file_player_->StopPlayingFile() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_STOP_PLAYOUT, kTraceError,
- "StopPlayingFile() couldnot stop playing file");
- return -1;
- }
-
- file_player_->RegisterModuleFileCallback(NULL);
- file_player_.reset();
- _filePlaying = false;
-
- return 0;
-}
-
-int TransmitMixer::IsPlayingFileAsMicrophone() const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::IsPlayingFileAsMicrophone()");
- return _filePlaying;
-}
-
-int TransmitMixer::StartRecordingMicrophone(const char* fileName,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingMicrophone(fileName=%s)",
- fileName);
-
- rtc::CritScope cs(&_critSect);
-
- if (_fileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingMicrophone() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
-
- if (codecInst != NULL && codecInst->channels > 2)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() invalid compression");
- return (-1);
- }
- if (codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- } else
- {
- format = kFileFormatCompressedFile;
- }
-
- // Destroy the old instance
- if (file_recorder_) {
- file_recorder_->RegisterModuleFileCallback(NULL);
- file_recorder_.reset();
- }
-
- file_recorder_ = FileRecorder::CreateFileRecorder(
- _fileRecorderId, (const FileFormats)format);
- if (!file_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() fileRecorder format isnot correct");
- return -1;
- }
-
- if (file_recorder_->StartRecordingAudioFile(
- fileName, (const CodecInst&)*codecInst, notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- file_recorder_->StopRecording();
- file_recorder_.reset();
- return -1;
- }
- file_recorder_->RegisterModuleFileCallback(this);
- _fileRecording = true;
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingMicrophone(OutStream* stream,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingMicrophone()");
-
- rtc::CritScope cs(&_critSect);
-
- if (_fileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingMicrophone() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() invalid compression");
- return (-1);
- }
- if (codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- } else
- {
- format = kFileFormatCompressedFile;
- }
-
- // Destroy the old instance
- if (file_recorder_) {
- file_recorder_->RegisterModuleFileCallback(NULL);
- file_recorder_.reset();
- }
-
- file_recorder_ = FileRecorder::CreateFileRecorder(
- _fileRecorderId, (const FileFormats)format);
- if (!file_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() fileRecorder format isnot correct");
- return -1;
- }
-
- if (file_recorder_->StartRecordingAudioFile(stream, *codecInst,
- notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- file_recorder_->StopRecording();
- file_recorder_.reset();
- return -1;
- }
-
- file_recorder_->RegisterModuleFileCallback(this);
- _fileRecording = true;
-
- return 0;
-}
-
-
-int TransmitMixer::StopRecordingMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StopRecordingMicrophone()");
-
- rtc::CritScope cs(&_critSect);
-
- if (!_fileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StopRecordingMicrophone() isnot recording");
- return 0;
- }
-
- if (file_recorder_->StopRecording() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording(), could not stop recording");
- return -1;
- }
- file_recorder_->RegisterModuleFileCallback(NULL);
- file_recorder_.reset();
- _fileRecording = false;
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingCall(const char* fileName,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingCall(fileName=%s)", fileName);
-
- if (_fileCallRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingCall() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingCall() invalid compression");
- return (-1);
- }
- if (codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- } else
- {
- format = kFileFormatCompressedFile;
- }
-
- rtc::CritScope cs(&_critSect);
-
- // Destroy the old instance
- if (file_call_recorder_) {
- file_call_recorder_->RegisterModuleFileCallback(NULL);
- file_call_recorder_.reset();
- }
-
- file_call_recorder_ = FileRecorder::CreateFileRecorder(
- _fileCallRecorderId, (const FileFormats)format);
- if (!file_call_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingCall() fileRecorder format isnot correct");
- return -1;
- }
-
- if (file_call_recorder_->StartRecordingAudioFile(
- fileName, (const CodecInst&)*codecInst, notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- file_call_recorder_->StopRecording();
- file_call_recorder_.reset();
- return -1;
- }
- file_call_recorder_->RegisterModuleFileCallback(this);
- _fileCallRecording = true;
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingCall(OutStream* stream,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingCall()");
-
- if (_fileCallRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingCall() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingCall() invalid compression");
- return (-1);
- }
- if (codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- } else
- {
- format = kFileFormatCompressedFile;
- }
-
- rtc::CritScope cs(&_critSect);
-
- // Destroy the old instance
- if (file_call_recorder_) {
- file_call_recorder_->RegisterModuleFileCallback(NULL);
- file_call_recorder_.reset();
- }
-
- file_call_recorder_ = FileRecorder::CreateFileRecorder(
- _fileCallRecorderId, (const FileFormats)format);
- if (!file_call_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingCall() fileRecorder format isnot correct");
- return -1;
- }
-
- if (file_call_recorder_->StartRecordingAudioFile(stream, *codecInst,
- notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- file_call_recorder_->StopRecording();
- file_call_recorder_.reset();
- return -1;
- }
-
- file_call_recorder_->RegisterModuleFileCallback(this);
- _fileCallRecording = true;
-
- return 0;
-}
-
-int TransmitMixer::StopRecordingCall()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StopRecordingCall()");
-
- if (!_fileCallRecording)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StopRecordingCall() file isnot recording");
- return -1;
- }
-
- rtc::CritScope cs(&_critSect);
-
- if (file_call_recorder_->StopRecording() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording(), could not stop recording");
- return -1;
- }
-
- file_call_recorder_->RegisterModuleFileCallback(NULL);
- file_call_recorder_.reset();
- _fileCallRecording = false;
-
- return 0;
-}
-
-void
-TransmitMixer::SetMixWithMicStatus(bool mix)
-{
- _mixFileWithMicrophone = mix;
-}
-
int8_t TransmitMixer::AudioLevel() const
{
// Speech + file level [0,9]
@@ -867,17 +269,6 @@
return _audioLevel.TotalDuration();
}
-bool TransmitMixer::IsRecordingCall()
-{
- return _fileCallRecording;
-}
-
-bool TransmitMixer::IsRecordingMic()
-{
- rtc::CritScope cs(&_critSect);
- return _fileRecording;
-}
-
void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
size_t samples_per_channel,
size_t num_channels,
@@ -901,79 +292,6 @@
&resampler_, &_audioFrame);
}
-int32_t TransmitMixer::RecordAudioToFile(
- uint32_t mixingFrequency)
-{
- rtc::CritScope cs(&_critSect);
- if (!file_recorder_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordAudioToFile() filerecorder doesnot"
- "exist");
- return -1;
- }
-
- if (file_recorder_->RecordAudioToFile(_audioFrame) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordAudioToFile() file recording"
- "failed");
- return -1;
- }
-
- return 0;
-}
-
-int32_t TransmitMixer::MixOrReplaceAudioWithFile(
- int mixingFrequency)
-{
- std::unique_ptr<int16_t[]> fileBuffer(new int16_t[640]);
-
- size_t fileSamples(0);
- {
- rtc::CritScope cs(&_critSect);
- if (!file_player_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::MixOrReplaceAudioWithFile()"
- "fileplayer doesnot exist");
- return -1;
- }
-
- if (file_player_->Get10msAudioFromFile(fileBuffer.get(), &fileSamples,
- mixingFrequency) == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::MixOrReplaceAudioWithFile() file"
- " mixing failed");
- return -1;
- }
- }
-
- assert(_audioFrame.samples_per_channel_ == fileSamples);
-
- if (_mixFileWithMicrophone)
- {
- // Currently file stream is always mono.
- // TODO(xians): Change the code when FilePlayer supports real stereo.
- MixWithSat(_audioFrame.mutable_data(),
- _audioFrame.num_channels_,
- fileBuffer.get(),
- 1,
- fileSamples);
- } else
- {
- // Replace ACM audio with file.
- // Currently file stream is always mono.
- // TODO(xians): Change the code when FilePlayer supports real stereo.
- _audioFrame.UpdateFrame(-1,
- 0xFFFFFFFF,
- fileBuffer.get(),
- fileSamples,
- mixingFrequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadUnknown,
- 1);
- }
- return 0;
-}
-
void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
int current_mic_level, bool key_pressed) {
if (audioproc_->set_stream_delay_ms(delay_ms) != 0) {
diff --git a/voice_engine/transmit_mixer.h b/voice_engine/transmit_mixer.h
index e480391..a04f92b 100644
--- a/voice_engine/transmit_mixer.h
+++ b/voice_engine/transmit_mixer.h
@@ -19,8 +19,6 @@
#include "modules/include/module_common_types.h"
#include "rtc_base/criticalsection.h"
#include "voice_engine/audio_level.h"
-#include "voice_engine/file_player.h"
-#include "voice_engine/file_recorder.h"
#include "voice_engine/include/voe_base.h"
#include "voice_engine/monitor_module.h"
#include "voice_engine/voice_engine_defines.h"
@@ -41,7 +39,7 @@
class MixedAudio;
class Statistics;
-class TransmitMixer : public FileCallback {
+class TransmitMixer {
public:
static int32_t Create(TransmitMixer*& mixer, uint32_t instanceId);
@@ -84,45 +82,6 @@
// 'virtual' to allow mocking.
virtual double GetTotalInputDuration() const;
- bool IsRecordingCall();
-
- bool IsRecordingMic();
-
- int StartPlayingFileAsMicrophone(const char* fileName,
- bool loop,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst);
-
- int StartPlayingFileAsMicrophone(InStream* stream,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst);
-
- int StopPlayingFileAsMicrophone();
-
- int IsPlayingFileAsMicrophone() const;
-
- int StartRecordingMicrophone(const char* fileName,
- const CodecInst* codecInst);
-
- int StartRecordingMicrophone(OutStream* stream,
- const CodecInst* codecInst);
-
- int StopRecordingMicrophone();
-
- int StartRecordingCall(const char* fileName, const CodecInst* codecInst);
-
- int StartRecordingCall(OutStream* stream, const CodecInst* codecInst);
-
- int StopRecordingCall();
-
- void SetMixWithMicStatus(bool mix);
-
int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
virtual ~TransmitMixer();
@@ -132,17 +91,6 @@
void OnPeriodicProcess();
#endif
- // FileCallback
- void PlayNotification(const int32_t id,
- const uint32_t durationMs);
-
- void RecordNotification(const int32_t id,
- const uint32_t durationMs);
-
- void PlayFileEnded(const int32_t id);
-
- void RecordFileEnded(const int32_t id);
-
// Virtual to allow mocking.
virtual void EnableStereoChannelSwapping(bool enable);
bool IsStereoChannelSwappingEnabled();
@@ -165,10 +113,6 @@
size_t nSamples,
size_t nChannels,
int samplesPerSec);
- int32_t RecordAudioToFile(uint32_t mixingFrequency);
-
- int32_t MixOrReplaceAudioWithFile(
- int mixingFrequency);
void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level,
bool key_pressed);
@@ -187,15 +131,6 @@
// owns
AudioFrame _audioFrame;
PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate
- std::unique_ptr<FilePlayer> file_player_;
- std::unique_ptr<FileRecorder> file_recorder_;
- std::unique_ptr<FileRecorder> file_call_recorder_;
- int _filePlayerId = 0;
- int _fileRecorderId = 0;
- int _fileCallRecorderId = 0;
- bool _filePlaying = false;
- bool _fileRecording = false;
- bool _fileCallRecording = false;
voe::AudioLevel _audioLevel;
// protect file instances and their variables in MixedParticipants()
rtc::CriticalSection _critSect;
@@ -209,7 +144,6 @@
#endif
int _instanceId = 0;
- bool _mixFileWithMicrophone = false;
uint32_t _captureLevel = 0;
bool stereo_codec_ = false;
bool swap_stereo_channels_ = false;
diff --git a/voice_engine/utility.cc b/voice_engine/utility.cc
index bd9ea8b..ecc3e22 100644
--- a/voice_engine/utility.cc
+++ b/voice_engine/utility.cc
@@ -88,41 +88,5 @@
}
}
-void MixWithSat(int16_t target[],
- size_t target_channel,
- const int16_t source[],
- size_t source_channel,
- size_t source_len) {
- RTC_DCHECK_GE(target_channel, 1);
- RTC_DCHECK_LE(target_channel, 2);
- RTC_DCHECK_GE(source_channel, 1);
- RTC_DCHECK_LE(source_channel, 2);
-
- if (target_channel == 2 && source_channel == 1) {
- // Convert source from mono to stereo.
- int32_t left = 0;
- int32_t right = 0;
- for (size_t i = 0; i < source_len; ++i) {
- left = source[i] + target[i * 2];
- right = source[i] + target[i * 2 + 1];
- target[i * 2] = WebRtcSpl_SatW32ToW16(left);
- target[i * 2 + 1] = WebRtcSpl_SatW32ToW16(right);
- }
- } else if (target_channel == 1 && source_channel == 2) {
- // Convert source from stereo to mono.
- int32_t temp = 0;
- for (size_t i = 0; i < source_len / 2; ++i) {
- temp = ((source[i * 2] + source[i * 2 + 1]) >> 1) + target[i];
- target[i] = WebRtcSpl_SatW32ToW16(temp);
- }
- } else {
- int32_t temp = 0;
- for (size_t i = 0; i < source_len; ++i) {
- temp = source[i] + target[i];
- target[i] = WebRtcSpl_SatW32ToW16(temp);
- }
- }
-}
-
} // namespace voe
} // namespace webrtc
diff --git a/voice_engine/utility.h b/voice_engine/utility.h
index c0c11ee..dc23e16 100644
--- a/voice_engine/utility.h
+++ b/voice_engine/utility.h
@@ -45,12 +45,6 @@
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
-void MixWithSat(int16_t target[],
- size_t target_channel,
- const int16_t source[],
- size_t source_channel,
- size_t source_len);
-
} // namespace voe
} // namespace webrtc
diff --git a/voice_engine/voe_base_impl.cc b/voice_engine/voe_base_impl.cc
index 8768b32..ec28e06 100644
--- a/voice_engine/voe_base_impl.cc
+++ b/voice_engine/voe_base_impl.cc
@@ -597,8 +597,7 @@
}
int32_t VoEBaseImpl::StopSend() {
- if (shared_->NumOfSendingChannels() == 0 &&
- !shared_->transmit_mixer()->IsRecordingMic()) {
+ if (shared_->NumOfSendingChannels() == 0) {
// Stop audio-device recording if no channel is recording
if (shared_->audio_device()->StopRecording() != 0) {
shared_->SetLastError(VE_CANNOT_STOP_RECORDING, kTraceError,
diff --git a/voice_engine/voe_file_impl.cc b/voice_engine/voe_file_impl.cc
deleted file mode 100644
index 9bf8c7a..0000000
--- a/voice_engine/voe_file_impl.cc
+++ /dev/null
@@ -1,466 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voice_engine/voe_file_impl.h"
-
-#include "system_wrappers/include/file_wrapper.h"
-#include "system_wrappers/include/trace.h"
-#include "voice_engine/channel.h"
-#include "voice_engine/include/voe_errors.h"
-#include "voice_engine/output_mixer.h"
-#include "voice_engine/transmit_mixer.h"
-#include "voice_engine/voice_engine_impl.h"
-
-namespace webrtc {
-
-VoEFile* VoEFile::GetInterface(VoiceEngine* voiceEngine) {
- if (NULL == voiceEngine) {
- return NULL;
- }
- VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
- s->AddRef();
- return s;
-}
-
-VoEFileImpl::VoEFileImpl(voe::SharedData* shared) : _shared(shared) {
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "VoEFileImpl::VoEFileImpl() - ctor");
-}
-
-VoEFileImpl::~VoEFileImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "VoEFileImpl::~VoEFileImpl() - dtor");
-}
-
-int VoEFileImpl::StartPlayingFileLocally(int channel,
- const char fileNameUTF8[1024],
- bool loop,
- FileFormats format,
- float volumeScaling,
- int startPointMs,
- int stopPointMs) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartPlayingFileLocally(channel=%d, fileNameUTF8[]=%s, "
- "loop=%d, format=%d, volumeScaling=%5.3f, startPointMs=%d,"
- " stopPointMs=%d)",
- channel, fileNameUTF8, loop, format, volumeScaling, startPointMs,
- stopPointMs);
- static_assert(1024 == FileWrapper::kMaxFileNameSize, "");
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StartPlayingFileLocally() failed to locate channel");
- return -1;
- }
-
- return channelPtr->StartPlayingFileLocally(fileNameUTF8, loop, format,
- startPointMs, volumeScaling,
- stopPointMs, NULL);
-}
-
-int VoEFileImpl::StartPlayingFileLocally(int channel,
- InStream* stream,
- FileFormats format,
- float volumeScaling,
- int startPointMs,
- int stopPointMs) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartPlayingFileLocally(channel=%d, stream, format=%d, "
- "volumeScaling=%5.3f, startPointMs=%d, stopPointMs=%d)",
- channel, format, volumeScaling, startPointMs, stopPointMs);
-
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StartPlayingFileLocally() failed to locate channel");
- return -1;
- }
-
- return channelPtr->StartPlayingFileLocally(stream, format, startPointMs,
- volumeScaling, stopPointMs, NULL);
-}
-
-int VoEFileImpl::StopPlayingFileLocally(int channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StopPlayingFileLocally()");
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StopPlayingFileLocally() failed to locate channel");
- return -1;
- }
- return channelPtr->StopPlayingFileLocally();
-}
-
-int VoEFileImpl::IsPlayingFileLocally(int channel) {
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StopPlayingFileLocally() failed to locate channel");
- return -1;
- }
- return channelPtr->IsPlayingFileLocally();
-}
-
-int VoEFileImpl::StartPlayingFileAsMicrophone(int channel,
- const char fileNameUTF8[1024],
- bool loop,
- bool mixWithMicrophone,
- FileFormats format,
- float volumeScaling) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartPlayingFileAsMicrophone(channel=%d, fileNameUTF8=%s, "
- "loop=%d, mixWithMicrophone=%d, format=%d, "
- "volumeScaling=%5.3f)",
- channel, fileNameUTF8, loop, mixWithMicrophone, format,
- volumeScaling);
- static_assert(1024 == FileWrapper::kMaxFileNameSize, "");
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- const uint32_t startPointMs(0);
- const uint32_t stopPointMs(0);
-
- if (channel == -1) {
- int res = _shared->transmit_mixer()->StartPlayingFileAsMicrophone(
- fileNameUTF8, loop, format, startPointMs, volumeScaling, stopPointMs,
- NULL);
- if (res) {
- WEBRTC_TRACE(
- kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartPlayingFileAsMicrophone() failed to start playing file");
- return (-1);
- } else {
- _shared->transmit_mixer()->SetMixWithMicStatus(mixWithMicrophone);
- return (0);
- }
- } else {
- // Add file after demultiplexing <=> affects one channel only
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StartPlayingFileAsMicrophone() failed to locate channel");
- return -1;
- }
-
- int res = channelPtr->StartPlayingFileAsMicrophone(
- fileNameUTF8, loop, format, startPointMs, volumeScaling, stopPointMs,
- NULL);
- if (res) {
- WEBRTC_TRACE(
- kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartPlayingFileAsMicrophone() failed to start playing file");
- return -1;
- } else {
- channelPtr->SetMixWithMicStatus(mixWithMicrophone);
- return 0;
- }
- }
-}
-
-int VoEFileImpl::StartPlayingFileAsMicrophone(int channel,
- InStream* stream,
- bool mixWithMicrophone,
- FileFormats format,
- float volumeScaling) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartPlayingFileAsMicrophone(channel=%d, stream,"
- " mixWithMicrophone=%d, format=%d, volumeScaling=%5.3f)",
- channel, mixWithMicrophone, format, volumeScaling);
-
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- const uint32_t startPointMs(0);
- const uint32_t stopPointMs(0);
-
- if (channel == -1) {
- int res = _shared->transmit_mixer()->StartPlayingFileAsMicrophone(
- stream, format, startPointMs, volumeScaling, stopPointMs, NULL);
- if (res) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartPlayingFileAsMicrophone() failed to start "
- "playing stream");
- return (-1);
- } else {
- _shared->transmit_mixer()->SetMixWithMicStatus(mixWithMicrophone);
- return (0);
- }
- } else {
- // Add file after demultiplexing <=> affects one channel only
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StartPlayingFileAsMicrophone() failed to locate channel");
- return -1;
- }
-
- int res = channelPtr->StartPlayingFileAsMicrophone(
- stream, format, startPointMs, volumeScaling, stopPointMs, NULL);
- if (res) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartPlayingFileAsMicrophone() failed to start "
- "playing stream");
- return -1;
- } else {
- channelPtr->SetMixWithMicStatus(mixWithMicrophone);
- return 0;
- }
- }
-}
-
-int VoEFileImpl::StopPlayingFileAsMicrophone(int channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StopPlayingFileAsMicrophone(channel=%d)", channel);
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1) {
- // Stop adding file before demultiplexing <=> affects all channels
- return _shared->transmit_mixer()->StopPlayingFileAsMicrophone();
- } else {
- // Stop adding file after demultiplexing <=> affects one channel only
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StopPlayingFileAsMicrophone() failed to locate channel");
- return -1;
- }
- return channelPtr->StopPlayingFileAsMicrophone();
- }
-}
-
-int VoEFileImpl::IsPlayingFileAsMicrophone(int channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "IsPlayingFileAsMicrophone(channel=%d)", channel);
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1) {
- return _shared->transmit_mixer()->IsPlayingFileAsMicrophone();
- } else {
- // Stop adding file after demultiplexing <=> affects one channel only
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "IsPlayingFileAsMicrophone() failed to locate channel");
- return -1;
- }
- return channelPtr->IsPlayingFileAsMicrophone();
- }
-}
-
-int VoEFileImpl::StartRecordingPlayout(int channel,
- const char* fileNameUTF8,
- CodecInst* compression,
- int maxSizeBytes) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingPlayout(channel=%d, fileNameUTF8=%s, "
- "compression, maxSizeBytes=%d)",
- channel, fileNameUTF8, maxSizeBytes);
- static_assert(1024 == FileWrapper::kMaxFileNameSize, "");
-
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1) {
- return _shared->output_mixer()->StartRecordingPlayout(fileNameUTF8,
- compression);
- } else {
- // Add file after demultiplexing <=> affects one channel only
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StartRecordingPlayout() failed to locate channel");
- return -1;
- }
- return channelPtr->StartRecordingPlayout(fileNameUTF8, compression);
- }
-}
-
-int VoEFileImpl::StartRecordingPlayout(int channel,
- OutStream* stream,
- CodecInst* compression) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingPlayout(channel=%d, stream, compression)",
- channel);
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1) {
- return _shared->output_mixer()->StartRecordingPlayout(stream, compression);
- } else {
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StartRecordingPlayout() failed to locate channel");
- return -1;
- }
- return channelPtr->StartRecordingPlayout(stream, compression);
- }
-}
-
-int VoEFileImpl::StopRecordingPlayout(int channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StopRecordingPlayout(channel=%d)", channel);
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1) {
- return _shared->output_mixer()->StopRecordingPlayout();
- } else {
- voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
- voe::Channel* channelPtr = ch.channel();
- if (channelPtr == NULL) {
- _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StopRecordingPlayout() failed to locate channel");
- return -1;
- }
- return channelPtr->StopRecordingPlayout();
- }
-}
-
-int VoEFileImpl::StartRecordingMicrophone(const char* fileNameUTF8,
- CodecInst* compression,
- int maxSizeBytes) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingMicrophone(fileNameUTF8=%s, compression, "
- "maxSizeBytes=%d)",
- fileNameUTF8, maxSizeBytes);
- static_assert(1024 == FileWrapper::kMaxFileNameSize, "");
-
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (_shared->transmit_mixer()->StartRecordingMicrophone(fileNameUTF8,
- compression)) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingMicrophone() failed to start recording");
- return -1;
- }
- if (!_shared->audio_device()->Recording()) {
- if (_shared->audio_device()->InitRecording() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingMicrophone() failed to initialize recording");
- return -1;
- }
- if (_shared->audio_device()->StartRecording() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingMicrophone() failed to start recording");
- return -1;
- }
- }
- return 0;
-}
-
-int VoEFileImpl::StartRecordingMicrophone(OutStream* stream,
- CodecInst* compression) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingMicrophone(stream, compression)");
-
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (_shared->transmit_mixer()->StartRecordingMicrophone(stream,
- compression) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingMicrophone() failed to start recording");
- return -1;
- }
- if (!_shared->audio_device()->Recording()) {
- if (_shared->audio_device()->InitRecording() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingMicrophone() failed to initialize recording");
- return -1;
- }
- if (_shared->audio_device()->StartRecording() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StartRecordingMicrophone() failed to start recording");
- return -1;
- }
- }
- return 0;
-}
-
-int VoEFileImpl::StopRecordingMicrophone() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StopRecordingMicrophone()");
- if (!_shared->statistics().Initialized()) {
- _shared->SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- int err = 0;
-
- // TODO(xians): consider removing Start/StopRecording() in
- // Start/StopRecordingMicrophone() if no channel is recording.
- if (_shared->NumOfSendingChannels() == 0 &&
- _shared->audio_device()->Recording()) {
- // Stop audio-device recording if no channel is recording
- if (_shared->audio_device()->StopRecording() != 0) {
- _shared->SetLastError(
- VE_CANNOT_STOP_RECORDING, kTraceError,
- "StopRecordingMicrophone() failed to stop recording");
- err = -1;
- }
- }
-
- if (_shared->transmit_mixer()->StopRecordingMicrophone() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
- "StopRecordingMicrophone() failed to stop recording to mixer");
- err = -1;
- }
-
- return err;
-}
-
-} // namespace webrtc
diff --git a/voice_engine/voe_file_impl.h b/voice_engine/voe_file_impl.h
deleted file mode 100644
index d3392a1..0000000
--- a/voice_engine/voe_file_impl.h
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef VOICE_ENGINE_VOE_FILE_IMPL_H_
-#define VOICE_ENGINE_VOE_FILE_IMPL_H_
-
-#include "voice_engine/include/voe_file.h"
-#include "voice_engine/shared_data.h"
-
-namespace webrtc {
-
-class VoEFileImpl : public VoEFile {
- public:
- // Playout file locally
-
- int StartPlayingFileLocally(int channel,
- const char fileNameUTF8[1024],
- bool loop = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0,
- int stopPointMs = 0) override;
-
- int StartPlayingFileLocally(int channel,
- InStream* stream,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0,
- int stopPointMs = 0) override;
-
- int StopPlayingFileLocally(int channel) override;
-
- int IsPlayingFileLocally(int channel) override;
-
- // Use file as microphone input
-
- int StartPlayingFileAsMicrophone(int channel,
- const char fileNameUTF8[1024],
- bool loop = false,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0) override;
-
- int StartPlayingFileAsMicrophone(int channel,
- InStream* stream,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0) override;
-
- int StopPlayingFileAsMicrophone(int channel) override;
-
- int IsPlayingFileAsMicrophone(int channel) override;
-
- // Record speaker signal to file
-
- int StartRecordingPlayout(int channel,
- const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1) override;
-
- int StartRecordingPlayout(int channel,
- OutStream* stream,
- CodecInst* compression = NULL) override;
-
- int StopRecordingPlayout(int channel) override;
-
- // Record microphone signal to file
-
- int StartRecordingMicrophone(const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1) override;
-
- int StartRecordingMicrophone(OutStream* stream,
- CodecInst* compression = NULL) override;
-
- int StopRecordingMicrophone() override;
-
- protected:
- VoEFileImpl(voe::SharedData* shared);
- ~VoEFileImpl() override;
-
- private:
- voe::SharedData* _shared;
-};
-
-} // namespace webrtc
-
-#endif // VOICE_ENGINE_VOE_FILE_IMPL_H_
diff --git a/voice_engine/voice_engine_impl.h b/voice_engine/voice_engine_impl.h
index fab4cdc..3a6fe27 100644
--- a/voice_engine/voice_engine_impl.h
+++ b/voice_engine/voice_engine_impl.h
@@ -17,7 +17,6 @@
#include "typedefs.h" // NOLINT(build/include)
#include "voice_engine/voe_base_impl.h"
#include "voice_engine/voe_codec_impl.h"
-#include "voice_engine/voe_file_impl.h"
#include "voice_engine/voe_network_impl.h"
#include "voice_engine/voe_rtp_rtcp_impl.h"
@@ -29,7 +28,6 @@
class VoiceEngineImpl : public voe::SharedData, // Must be the first base class
public VoiceEngine,
public VoECodecImpl,
- public VoEFileImpl,
public VoENetworkImpl,
public VoERTP_RTCPImpl,
public VoEBaseImpl {
@@ -37,7 +35,6 @@
VoiceEngineImpl()
: SharedData(),
VoECodecImpl(this),
- VoEFileImpl(this),
VoENetworkImpl(this),
VoERTP_RTCPImpl(this),
VoEBaseImpl(this),
