Delete some unused AudioCodingModule methods
Methods deleted:
ReceiveFrequency, PlayoutFrequency, ReceiveCodec,
SetMinimumPlayoutDelay, SetMaximumPlayoutDelay,
SetBaseMinimumPlayoutDelayMs, GetBaseMinimumPlayoutDelayMs,
PlayoutTimestamp, FilteredCurrentDelayMs, TargetDelayMs.
Became unused with cl
https://webrtc-review.googlesource.com/c/src/+/111504
Bug: None
Change-Id: Ie50e8e86a622661c3daa9db83a2e66489dcd2d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148071
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28918}
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index b0421f5..ef2e0e5 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -74,38 +74,13 @@
// Initialize receiver, resets codec database etc.
int InitializeReceiver() override;
- // Get current receive frequency.
- int ReceiveFrequency() const override;
-
- // Get current playout frequency.
- int PlayoutFrequency() const override;
-
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
- // Get current received codec.
- absl::optional<std::pair<int, SdpAudioFormat>> ReceiveCodec() const override;
-
// Incoming packet from network parsed and ready for decode.
int IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
const RTPHeader& rtp_info) override;
- // Minimum playout delay.
- int SetMinimumPlayoutDelay(int time_ms) override;
-
- // Maximum playout delay.
- int SetMaximumPlayoutDelay(int time_ms) override;
-
- bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
-
- int GetBaseMinimumPlayoutDelayMs() const override;
-
- absl::optional<uint32_t> PlayoutTimestamp() override;
-
- int FilteredCurrentDelayMs() const override;
-
- int TargetDelayMs() const override;
-
// Get 10 milliseconds of raw audio data to play out, and
// automatic resample to the requested frequency if > 0.
int PlayoutData10Ms(int desired_freq_hz,
@@ -605,30 +580,12 @@
return 0;
}
-// Get current receive frequency.
-int AudioCodingModuleImpl::ReceiveFrequency() const {
- const auto last_packet_sample_rate = receiver_.last_packet_sample_rate_hz();
- return last_packet_sample_rate ? *last_packet_sample_rate
- : receiver_.last_output_sample_rate_hz();
-}
-
-// Get current playout frequency.
-int AudioCodingModuleImpl::PlayoutFrequency() const {
- return receiver_.last_output_sample_rate_hz();
-}
-
void AudioCodingModuleImpl::SetReceiveCodecs(
const std::map<int, SdpAudioFormat>& codecs) {
rtc::CritScope lock(&acm_crit_sect_);
receiver_.SetCodecs(codecs);
}
-absl::optional<std::pair<int, SdpAudioFormat>>
-AudioCodingModuleImpl::ReceiveCodec() const {
- rtc::CritScope lock(&acm_crit_sect_);
- return receiver_.LastDecoder();
-}
-
// Incoming packet from network parsed and ready for decode.
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
@@ -639,32 +596,6 @@
rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
}
-// Minimum playout delay (Used for lip-sync).
-int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
- if ((time_ms < 0) || (time_ms > 10000)) {
- RTC_LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
- return -1;
- }
- return receiver_.SetMinimumDelay(time_ms);
-}
-
-int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
- if ((time_ms < 0) || (time_ms > 10000)) {
- RTC_LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
- return -1;
- }
- return receiver_.SetMaximumDelay(time_ms);
-}
-
-bool AudioCodingModuleImpl::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
- // All necessary validation happens on NetEq level.
- return receiver_.SetBaseMinimumDelayMs(delay_ms);
-}
-
-int AudioCodingModuleImpl::GetBaseMinimumPlayoutDelayMs() const {
- return receiver_.GetBaseMinimumDelayMs();
-}
-
// Get 10 milliseconds of raw audio data to play out.
// Automatic resample to the requested frequency.
int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
@@ -696,18 +627,6 @@
return 0;
}
-absl::optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
- return receiver_.GetPlayoutTimestamp();
-}
-
-int AudioCodingModuleImpl::FilteredCurrentDelayMs() const {
- return receiver_.FilteredCurrentDelayMs();
-}
-
-int AudioCodingModuleImpl::TargetDelayMs() const {
- return receiver_.TargetDelayMs();
-}
-
bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
if (!encoder_stack_) {
RTC_LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 177ce3b..df91c0b 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -185,42 +185,11 @@
//
virtual int32_t InitializeReceiver() = 0;
- ///////////////////////////////////////////////////////////////////////////
- // int32_t ReceiveFrequency()
- // Get sampling frequency of the last received payload.
- //
- // Return value:
- // non-negative the sampling frequency in Hertz.
- // -1 if an error has occurred.
- //
- virtual int32_t ReceiveFrequency() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // int32_t PlayoutFrequency()
- // Get sampling frequency of audio played out.
- //
- // Return value:
- // the sampling frequency in Hertz.
- //
- virtual int32_t PlayoutFrequency() const = 0;
-
// Replace any existing decoders with the given payload type -> decoder map.
virtual void SetReceiveCodecs(
const std::map<int, SdpAudioFormat>& codecs) = 0;
///////////////////////////////////////////////////////////////////////////
- // absl::optional<std::pair<int, SdpAudioFormat>> ReceiveCodec()
- // Get the codec info associated with last received payload.
- //
- // Return value:
- // A payload type and SdpAudioFormat describing the format associated with
- // the last received payload.
- // An empty Optional if no payload has yet been received.
- //
- virtual absl::optional<std::pair<int, SdpAudioFormat>> ReceiveCodec()
- const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
// int32_t IncomingPacket()
// Call this function to insert a parsed RTP packet into ACM.
//
@@ -239,66 +208,6 @@
const RTPHeader& rtp_header) = 0;
///////////////////////////////////////////////////////////////////////////
- // int SetMinimumPlayoutDelay()
- // Set a minimum for the playout delay, used for lip-sync. NetEq maintains
- // such a delay unless channel condition yields to a higher delay.
- //
- // Input:
- // -time_ms : minimum delay in milliseconds.
- //
- // Return value:
- // -1 if failed to set the delay,
- // 0 if the minimum delay is set.
- //
- virtual int SetMinimumPlayoutDelay(int time_ms) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // int SetMaximumPlayoutDelay()
- // Set a maximum for the playout delay
- //
- // Input:
- // -time_ms : maximum delay in milliseconds.
- //
- // Return value:
- // -1 if failed to set the delay,
- // 0 if the maximum delay is set.
- //
- virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
-
- // Sets a base minimum for the playout delay. Base minimum delay sets lower
- // bound minimum delay value which is set via SetMinimumPlayoutDelay.
- //
- // Returns true if value was successfully set, false overwise.
- virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
-
- // Returns current value of base minimum delay in milliseconds.
- virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // int32_t PlayoutTimestamp()
- // The send timestamp of an RTP packet is associated with the decoded
- // audio of the packet in question. This function returns the timestamp of
- // the latest audio obtained by calling PlayoutData10ms(), or empty if no
- // valid timestamp is available.
- //
- virtual absl::optional<uint32_t> PlayoutTimestamp() = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // int FilteredCurrentDelayMs()
- // Returns the current total delay from NetEq (packet buffer and sync buffer)
- // in ms, with smoothing applied to even out short-time fluctuations due to
- // jitter. The packet buffer part of the delay is not updated during DTX/CNG
- // periods.
- //
- virtual int FilteredCurrentDelayMs() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // int FilteredCurrentDelayMs()
- // Returns the current target delay for NetEq in ms.
- //
- virtual int TargetDelayMs() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutData10Ms(
// Get 10 milliseconds of raw audio data for playout, at the given sampling
// frequency. ACM will perform a resampling if required.
diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
index ae6c2b7..b85c8a0 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -186,9 +186,6 @@
Run10ms();
}
- EXPECT_TRUE(_acmA->ReceiveCodec());
- EXPECT_TRUE(_acmB->ReceiveCodec());
-
_inFileA.Close();
_outFileA.Close();
_outFileB.Close();
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 2fb59d1..77a2e5a 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -12,6 +12,7 @@
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/include/module_common_types.h"
@@ -23,19 +24,16 @@
class TargetDelayTest : public ::testing::Test {
protected:
TargetDelayTest()
- : acm_(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {}
+ : receiver_(
+ AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())) {}
~TargetDelayTest() {}
void SetUp() {
- EXPECT_TRUE(acm_.get() != NULL);
-
- ASSERT_EQ(0, acm_->InitializeReceiver());
constexpr int pltype = 108;
std::map<int, SdpAudioFormat> receive_codecs = {
{pltype, {"L16", kSampleRateHz, 1}}};
- acm_->SetReceiveCodecs(receive_codecs);
+ receiver_.SetCodecs(receive_codecs);
rtp_header_.payloadType = pltype;
rtp_header_.timestamp = 0;
@@ -99,8 +97,9 @@
void Push() {
rtp_header_.timestamp += kFrameSizeSamples;
rtp_header_.sequenceNumber++;
- ASSERT_EQ(
- 0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, rtp_header_));
+ ASSERT_EQ(0, receiver_.InsertPacket(rtp_header_,
+ rtc::ArrayView<const uint8_t>(
+ payload_, kFrameSizeSamples * 2)));
}
// Pull audio equivalent to the amount of audio in one RTP packet.
@@ -108,7 +107,7 @@
AudioFrame frame;
bool muted;
for (int k = 0; k < kNum10msPerFrame; ++k) { // Pull one frame.
- ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame, &muted));
+ ASSERT_EQ(0, receiver_.GetAudio(-1, &frame, &muted));
ASSERT_FALSE(muted);
// Had to use ASSERT_TRUE, ASSERT_EQ generated error.
ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
@@ -135,20 +134,20 @@
}
int SetMinimumDelay(int delay_ms) {
- return acm_->SetMinimumPlayoutDelay(delay_ms);
+ return receiver_.SetMinimumDelay(delay_ms);
}
int SetMaximumDelay(int delay_ms) {
- return acm_->SetMaximumPlayoutDelay(delay_ms);
+ return receiver_.SetMaximumDelay(delay_ms);
}
int GetCurrentOptimalDelayMs() {
NetworkStatistics stats;
- acm_->GetNetworkStatistics(&stats);
+ receiver_.GetNetworkStatistics(&stats);
return stats.preferredBufferSize;
}
- std::unique_ptr<AudioCodingModule> acm_;
+ acm2::AcmReceiver receiver_;
RTPHeader rtp_header_;
uint8_t payload_[kPayloadLenBytes];
};
