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Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/)

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Reason for revert:
Breaks Chromium FYI content_browsertest on all platforms. The testcase that fails is WebRtcAecDumpBrowserTest.CallWithAecDump.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/19388

Sample output:
[ RUN      ] WebRtcAecDumpBrowserTest.CallWithAecDump
Xlib:  extension "RANDR" missing on display ":9".
[4:14:0722/211548:1282124453:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: ISAC/48000/1 (105)
[4:14:0722/211548:1282124593:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMU/8000/2 (110)
[4:14:0722/211548:1282124700:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMA/8000/2 (118)
[4:14:0722/211548:1282124815:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: G722/8000/2 (119)
[19745:19745:0722/211548:1282133667:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
[19745:19745:0722/211548:1282136892:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
../../content/test/webrtc_content_browsertest_base.cc:62: Failure
Value of: ExecuteScriptAndExtractString( shell()->web_contents(), javascript, &result)
  Actual: false
Expected: true
Failed to execute javascript call({video: true, audio: true});.
From javascript: (nothing)
When executing 'call({video: true, audio: true});'
../../content/test/webrtc_content_browsertest_base.cc:75: Failure
Failed
../../content/browser/media/webrtc_aecdump_browsertest.cc:26: Failure
Expected: (base::kNullProcessId) != (*id), actual: 0 vs 0
../../content/browser/media/webrtc_aecdump_browsertest.cc:95: Failure
Value of: GetRenderProcessHostId(&render_process_id)
  Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:99: Failure
Value of: base::PathExists(dump_file)
  Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:101: Failure
Value of: base::GetFileSize(dump_file, &file_size)
  Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:102: Failure
Expected: (file_size) > (0), actual: 0 vs 0
[  FAILED  ] WebRtcAecDumpBrowserTest.CallWithAecDump, where TypeParam =  and GetParam() =  (361 ms)

Original issue's description:
> Allow more than 2 input channels in AudioProcessing.
>
> The number of output channels is constrained to be equal to either 1 or the
> number of input channels.
>
> R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
>
> Committed: c204754b7a

TBR=andrew@webrtc.org,aluebs@webrtc.org,ajm@chromium.org,pbos@chromium.org,pbos@webrtc.org,mgraczyk@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1253573005

Cr-Commit-Position: refs/heads/master@{#9621}
This commit is contained in:
magjed
2015-07-23 04:30:06 -07:00
committed by Commit bot
parent b21fd94ece
commit 64e753c399
13 changed files with 371 additions and 708 deletions
Download Patch File Download Diff File Expand all files Collapse all files

73
webrtc/modules/audio_processing/audio_processing_impl.h
View File

@ -13,7 +13,6 @@
#include <list>
#include <string>
#include <vector>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
@ -48,6 +47,42 @@ class Event;
} // namespace audioproc
#endif
class AudioRate {
public:
explicit AudioRate(int sample_rate_hz) { set(sample_rate_hz); }
virtual ~AudioRate() {}
void set(int rate) {
rate_ = rate;
samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
}
int rate() const { return rate_; }
int samples_per_channel() const { return samples_per_channel_; }
private:
int rate_;
int samples_per_channel_;
};
class AudioFormat : public AudioRate {
public:
AudioFormat(int sample_rate_hz, int num_channels)
: AudioRate(sample_rate_hz),
num_channels_(num_channels) {}
virtual ~AudioFormat() {}
void set(int rate, int num_channels) {
AudioRate::set(rate);
num_channels_ = num_channels;
}
int num_channels() const { return num_channels_; }
private:
int num_channels_;
};
class AudioProcessingImpl : public AudioProcessing {
public:
explicit AudioProcessingImpl(const Config& config);
@ -64,7 +99,6 @@ class AudioProcessingImpl : public AudioProcessing {
ChannelLayout input_layout,
ChannelLayout output_layout,
ChannelLayout reverse_layout) override;
int Initialize(const ProcessingConfig& processing_config) override;
void SetExtraOptions(const Config& config) override;
int set_sample_rate_hz(int rate) override;
int input_sample_rate_hz() const override;
@ -84,17 +118,11 @@ class AudioProcessingImpl : public AudioProcessing {
int output_sample_rate_hz,
ChannelLayout output_layout,
float* const* dest) override;
int ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) override;
int AnalyzeReverseStream(AudioFrame* frame) override;
int AnalyzeReverseStream(const float* const* data,
int samples_per_channel,
int sample_rate_hz,
ChannelLayout layout) override;
int AnalyzeReverseStream(const float* const* data,
const StreamConfig& reverse_config) override;
int set_stream_delay_ms(int delay) override;
int stream_delay_ms() const override;
bool was_stream_delay_set() const override;
@ -120,9 +148,19 @@ class AudioProcessingImpl : public AudioProcessing {
virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
private:
int InitializeLocked(const ProcessingConfig& config)
int InitializeLocked(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
int MaybeInitializeLocked(const ProcessingConfig& config)
int MaybeInitializeLocked(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
@ -159,14 +197,13 @@ class AudioProcessingImpl : public AudioProcessing {
std::string event_str_; // Memory for protobuf serialization.
#endif
// Format of processing streams at input/output call sites.
ProcessingConfig api_format_;
// Only the rate and samples fields of fwd_proc_format_ are used because the
// forward processing number of channels is mutable and is tracked by the
// capture_audio_.
StreamConfig fwd_proc_format_;
StreamConfig rev_proc_format_;
AudioFormat fwd_in_format_;
// This one is an AudioRate, because the forward processing number of channels
// is mutable and is tracked by the capture_audio_.
AudioRate fwd_proc_format_;
AudioFormat fwd_out_format_;
AudioFormat rev_in_format_;
AudioFormat rev_proc_format_;
int split_rate_;
int stream_delay_ms_;