Replace Resampler with PushResampler in transmit_mixer.
* VoE can now exchange 44.1 kHz audio with AudioDevice.
* Changes still required in AudioDevice to remove the 44 kHz workarounds and
enable native 44.1 kHz.
BUG=webrtc:1395
TESTED=voe_cmd_test loopback running through codecs using all combinations of {8, 16, 32} kHz and {1, 2} channels, and Opus (48 kHz, stereo)
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1373004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3930 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 4e6e6d6..ea35afb 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "transmit_mixer.h"
+#include "webrtc/voice_engine/transmit_mixer.h"
-#include "audio_frame_operations.h"
-#include "channel.h"
-#include "channel_manager.h"
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "statistics.h"
-#include "trace.h"
-#include "utility.h"
-#include "voe_base_impl.h"
-#include "voe_external_media.h"
+#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/voice_engine/channel.h"
+#include "webrtc/voice_engine/channel_manager.h"
+#include "webrtc/voice_engine/include/voe_external_media.h"
+#include "webrtc/voice_engine/statistics.h"
+#include "webrtc/voice_engine/utility.h"
+#include "webrtc/voice_engine/voe_base_impl.h"
#define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a))
@@ -1155,60 +1155,55 @@
int TransmitMixer::GenerateAudioFrame(const int16_t audio[],
int samples_per_channel,
int num_channels,
- int sample_rate_hz)
-{
- int destination_rate;
- int num_codec_channels;
- GetSendCodecInfo(&destination_rate, &num_codec_channels);
+ int sample_rate_hz) {
+ int destination_rate;
+ int num_codec_channels;
+ GetSendCodecInfo(&destination_rate, &num_codec_channels);
- // Never upsample the capture signal here. This should be done at the
- // end of the send chain.
- destination_rate = std::min(destination_rate, sample_rate_hz);
- stereo_codec_ = num_codec_channels == 2;
+ // Never upsample the capture signal here. This should be done at the
+ // end of the send chain.
+ destination_rate = std::min(destination_rate, sample_rate_hz);
+ stereo_codec_ = num_codec_channels == 2;
- const int16_t* audio_ptr = audio;
- int16_t mono_audio[kMaxMonoDeviceDataSizeSamples];
- assert(samples_per_channel <= kMaxMonoDeviceDataSizeSamples);
- // If no stereo codecs are in use, we downmix a stereo stream from the
- // device early in the chain, before resampling.
- if (num_channels == 2 && !stereo_codec_) {
- AudioFrameOperations::StereoToMono(audio, samples_per_channel,
- mono_audio);
- audio_ptr = mono_audio;
- num_channels = 1;
- }
+ const int16_t* audio_ptr = audio;
+ int16_t mono_audio[kMaxMonoDeviceDataSizeSamples];
+ assert(samples_per_channel <= kMaxMonoDeviceDataSizeSamples);
+ // If no stereo codecs are in use, we downmix a stereo stream from the
+ // device early in the chain, before resampling.
+ if (num_channels == 2 && !stereo_codec_) {
+ AudioFrameOperations::StereoToMono(audio, samples_per_channel,
+ mono_audio);
+ audio_ptr = mono_audio;
+ num_channels = 1;
+ }
- ResamplerType resampler_type = (num_channels == 1) ?
- kResamplerSynchronous : kResamplerSynchronousStereo;
+ if (resampler_.InitializeIfNeeded(sample_rate_hz,
+ destination_rate,
+ num_channels) != 0) {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::GenerateAudioFrame() unable to resample");
+ return -1;
+ }
- if (_audioResampler.ResetIfNeeded(sample_rate_hz,
- destination_rate,
- resampler_type) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::GenerateAudioFrame() unable to resample");
- return -1;
- }
- if (_audioResampler.Push(audio_ptr,
- samples_per_channel * num_channels,
- _audioFrame.data_,
- AudioFrame::kMaxDataSizeSamples,
- _audioFrame.samples_per_channel_) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::GenerateAudioFrame() resampling failed");
- return -1;
- }
+ int out_length = resampler_.Resample(audio_ptr,
+ samples_per_channel * num_channels,
+ _audioFrame.data_,
+ AudioFrame::kMaxDataSizeSamples);
+ if (out_length == -1) {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::GenerateAudioFrame() resampling failed");
+ return -1;
+ }
- _audioFrame.samples_per_channel_ /= num_channels;
- _audioFrame.id_ = _instanceId;
- _audioFrame.timestamp_ = -1;
- _audioFrame.sample_rate_hz_ = destination_rate;
- _audioFrame.speech_type_ = AudioFrame::kNormalSpeech;
- _audioFrame.vad_activity_ = AudioFrame::kVadUnknown;
- _audioFrame.num_channels_ = num_channels;
+ _audioFrame.samples_per_channel_ = out_length / num_channels;
+ _audioFrame.id_ = _instanceId;
+ _audioFrame.timestamp_ = -1;
+ _audioFrame.sample_rate_hz_ = destination_rate;
+ _audioFrame.speech_type_ = AudioFrame::kNormalSpeech;
+ _audioFrame.vad_activity_ = AudioFrame::kVadUnknown;
+ _audioFrame.num_channels_ = num_channels;
- return 0;
+ return 0;
}
int32_t TransmitMixer::RecordAudioToFile(
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index f01362f..c98f640 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -11,16 +11,15 @@
#ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
#define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
-#include "common_types.h"
-#include "voe_base.h"
-#include "file_player.h"
-#include "file_recorder.h"
-#include "level_indicator.h"
-#include "module_common_types.h"
-#include "monitor_module.h"
-#include "resampler.h"
-#include "voice_engine_defines.h"
-
+#include "webrtc/common_audio/resampler/include/push_resampler.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/utility/interface/file_player.h"
+#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/level_indicator.h"
+#include "webrtc/voice_engine/monitor_module.h"
+#include "webrtc/voice_engine/voice_engine_defines.h"
namespace webrtc {
@@ -192,7 +191,7 @@
// owns
MonitorModule _monitorModule;
AudioFrame _audioFrame;
- Resampler _audioResampler; // ADM sample rate -> mixing rate
+ PushResampler resampler_; // ADM sample rate -> mixing rate
FilePlayer* _filePlayerPtr;
FileRecorder* _fileRecorderPtr;
FileRecorder* _fileCallRecorderPtr;
