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
This commit is contained in:
andrew@webrtc.org
parent
342353780d
commit
28e82bfec6
@ -8,19 +8,19 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "transmit_mixer.h"
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#include "webrtc/voice_engine/transmit_mixer.h"
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#include "audio_frame_operations.h"
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#include "channel.h"
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#include "channel_manager.h"
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#include "critical_section_wrapper.h"
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#include "event_wrapper.h"
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#include "statistics.h"
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#include "trace.h"
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#include "utility.h"
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#include "voe_base_impl.h"
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#include "voe_external_media.h"
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#include "webrtc/modules/utility/interface/audio_frame_operations.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/event_wrapper.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#include "webrtc/voice_engine/channel.h"
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#include "webrtc/voice_engine/channel_manager.h"
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#include "webrtc/voice_engine/include/voe_external_media.h"
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#include "webrtc/voice_engine/statistics.h"
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#include "webrtc/voice_engine/utility.h"
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#include "webrtc/voice_engine/voe_base_impl.h"
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#define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a))
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@ -1155,60 +1155,55 @@ bool TransmitMixer::IsRecordingMic()
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int TransmitMixer::GenerateAudioFrame(const int16_t audio[],
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int samples_per_channel,
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int num_channels,
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int sample_rate_hz)
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{
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int destination_rate;
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int num_codec_channels;
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GetSendCodecInfo(&destination_rate, &num_codec_channels);
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int sample_rate_hz) {
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int destination_rate;
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int num_codec_channels;
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GetSendCodecInfo(&destination_rate, &num_codec_channels);
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// Never upsample the capture signal here. This should be done at the
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// end of the send chain.
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destination_rate = std::min(destination_rate, sample_rate_hz);
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stereo_codec_ = num_codec_channels == 2;
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// Never upsample the capture signal here. This should be done at the
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// end of the send chain.
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destination_rate = std::min(destination_rate, sample_rate_hz);
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stereo_codec_ = num_codec_channels == 2;
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const int16_t* audio_ptr = audio;
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int16_t mono_audio[kMaxMonoDeviceDataSizeSamples];
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assert(samples_per_channel <= kMaxMonoDeviceDataSizeSamples);
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// If no stereo codecs are in use, we downmix a stereo stream from the
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// device early in the chain, before resampling.
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if (num_channels == 2 && !stereo_codec_) {
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AudioFrameOperations::StereoToMono(audio, samples_per_channel,
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mono_audio);
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audio_ptr = mono_audio;
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num_channels = 1;
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}
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const int16_t* audio_ptr = audio;
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int16_t mono_audio[kMaxMonoDeviceDataSizeSamples];
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assert(samples_per_channel <= kMaxMonoDeviceDataSizeSamples);
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// If no stereo codecs are in use, we downmix a stereo stream from the
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// device early in the chain, before resampling.
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if (num_channels == 2 && !stereo_codec_) {
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AudioFrameOperations::StereoToMono(audio, samples_per_channel,
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mono_audio);
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audio_ptr = mono_audio;
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num_channels = 1;
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}
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ResamplerType resampler_type = (num_channels == 1) ?
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kResamplerSynchronous : kResamplerSynchronousStereo;
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if (resampler_.InitializeIfNeeded(sample_rate_hz,
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destination_rate,
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num_channels) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
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"TransmitMixer::GenerateAudioFrame() unable to resample");
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return -1;
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}
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if (_audioResampler.ResetIfNeeded(sample_rate_hz,
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destination_rate,
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resampler_type) != 0)
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{
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WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
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"TransmitMixer::GenerateAudioFrame() unable to resample");
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return -1;
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}
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if (_audioResampler.Push(audio_ptr,
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samples_per_channel * num_channels,
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_audioFrame.data_,
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AudioFrame::kMaxDataSizeSamples,
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_audioFrame.samples_per_channel_) == -1)
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{
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WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
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"TransmitMixer::GenerateAudioFrame() resampling failed");
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return -1;
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}
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int out_length = resampler_.Resample(audio_ptr,
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samples_per_channel * num_channels,
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_audioFrame.data_,
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AudioFrame::kMaxDataSizeSamples);
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if (out_length == -1) {
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WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
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"TransmitMixer::GenerateAudioFrame() resampling failed");
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return -1;
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}
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_audioFrame.samples_per_channel_ /= num_channels;
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_audioFrame.id_ = _instanceId;
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_audioFrame.timestamp_ = -1;
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_audioFrame.sample_rate_hz_ = destination_rate;
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_audioFrame.speech_type_ = AudioFrame::kNormalSpeech;
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_audioFrame.vad_activity_ = AudioFrame::kVadUnknown;
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_audioFrame.num_channels_ = num_channels;
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_audioFrame.samples_per_channel_ = out_length / num_channels;
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_audioFrame.id_ = _instanceId;
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_audioFrame.timestamp_ = -1;
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_audioFrame.sample_rate_hz_ = destination_rate;
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_audioFrame.speech_type_ = AudioFrame::kNormalSpeech;
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_audioFrame.vad_activity_ = AudioFrame::kVadUnknown;
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_audioFrame.num_channels_ = num_channels;
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return 0;
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return 0;
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}
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int32_t TransmitMixer::RecordAudioToFile(
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@ -11,16 +11,15 @@
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#ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
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#define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
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#include "common_types.h"
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#include "voe_base.h"
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#include "file_player.h"
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#include "file_recorder.h"
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#include "level_indicator.h"
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#include "module_common_types.h"
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#include "monitor_module.h"
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#include "resampler.h"
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#include "voice_engine_defines.h"
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#include "webrtc/common_audio/resampler/include/push_resampler.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/modules/utility/interface/file_player.h"
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#include "webrtc/modules/utility/interface/file_recorder.h"
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/voice_engine/level_indicator.h"
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#include "webrtc/voice_engine/monitor_module.h"
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#include "webrtc/voice_engine/voice_engine_defines.h"
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namespace webrtc {
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@ -192,7 +191,7 @@ private:
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// owns
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MonitorModule _monitorModule;
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AudioFrame _audioFrame;
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Resampler _audioResampler; // ADM sample rate -> mixing rate
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PushResampler resampler_; // ADM sample rate -> mixing rate
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FilePlayer* _filePlayerPtr;
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FileRecorder* _fileRecorderPtr;
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FileRecorder* _fileCallRecorderPtr;
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