diff options
Diffstat (limited to 'audio/audio_hw.c')
| -rw-r--r-- | audio/audio_hw.c | 88 |
1 files changed, 78 insertions, 10 deletions
diff --git a/audio/audio_hw.c b/audio/audio_hw.c index 403e5f3..aea5677 100644 --- a/audio/audio_hw.c +++ b/audio/audio_hw.c @@ -116,20 +116,70 @@ #define PORT_SPDIF 9 #define PORT_HDMI 0 -/* constraint imposed by ABE: all period sizes must be multiples of 24 */ +/* User serviceable */ +/* #define to use mmap no-irq mode for playback, #undef for non-mmap irq mode */ +#undef PLAYBACK_MMAP // was #define +/* short period (aka low latency) in milliseconds */ +#define SHORT_PERIOD_MS 4 // was 22 +/* long period (aka low power or deep buffer) in milliseconds */ +#define LONG_PERIOD_MS 308 + +/* Constraint imposed by ABE: for playback, all period sizes must be multiples of 24 frames + * = 500 us at 48 kHz. It seems to be either 48 or 96 for capture, or maybe it is because the + * limitation is actually a min number of bytes which translates to a different amount of frames + * according to the number of channels. + */ #define ABE_BASE_FRAME_COUNT 24 + +/* Derived from MM_FULL_POWER_SAMPLING_RATE=48000 and ABE_BASE_FRAME_COUNT=24 */ +#define MULTIPLIER_FACTOR 2 + /* number of base blocks in a short period (low latency) */ -#define SHORT_PERIOD_MULTIPLIER 44 /* 22 ms */ +#define SHORT_PERIOD_MULTIPLIER (SHORT_PERIOD_MS * MULTIPLIER_FACTOR) /* number of frames per short period (low latency) */ #define SHORT_PERIOD_SIZE (ABE_BASE_FRAME_COUNT * SHORT_PERIOD_MULTIPLIER) -/* number of short periods in a long period (low power) */ -#define LONG_PERIOD_MULTIPLIER 14 /* 308 ms */ + +/* number of short periods in a long period */ +#define LONG_PERIOD_MULTIPLIER (LONG_PERIOD_MS / SHORT_PERIOD_MS) /* number of frames per long period (low power) */ #define LONG_PERIOD_SIZE (SHORT_PERIOD_SIZE * LONG_PERIOD_MULTIPLIER) /* number of periods for low power playback */ #define PLAYBACK_LONG_PERIOD_COUNT 2 -/* number of pseudo periods for low latency playback */ +/* Number of pseudo periods for low latency playback. + * These are called "pseudo" periods in that they are not known as periods by ALSA. + * Formerly, ALSA was configured in MMAP mode with 2 large periods, and this + * number was set to 4 (2 didn't work). + * The short periods size and count were only known by the audio HAL. + * Now for low latency, we are using non-MMAP mode and can set this to 2. + */ +#ifdef PLAYBACK_MMAP #define PLAYBACK_SHORT_PERIOD_COUNT 4 +#else +#define PLAYBACK_SHORT_PERIOD_COUNT 2 +#endif + +/* write function */ +#ifdef PLAYBACK_MMAP +#define PCM_WRITE pcm_mmap_write +#else +#define PCM_WRITE pcm_write +#endif + +/* User serviceable */ +#define CAPTURE_PERIOD_MS 22 + +/* Number of frames per period for capture. This cannot be reduced below 96. + * Possibly related to the following rule in sound/soc/omap/omap-pcm.c: + * ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 384); + * (where 96 * 4 = 384) + * The only constraints I can find are periods_min = 2, period_bytes_min = 32. + * If you define RULES_DEBUG in sound/core/pcm_native.c, you can see which rule + * caused capture to fail. + * Decoupling playback and capture period size may have impacts on echo canceler behavior: + * to be verified. Currently 96 = 4 x 24 but it could be changed without noticing + * if we use separate defines. + */ +#define CAPTURE_PERIOD_SIZE (ABE_BASE_FRAME_COUNT * CAPTURE_PERIOD_MS * MULTIPLIER_FACTOR) /* number of periods for capture */ #define CAPTURE_PERIOD_COUNT 2 /* minimum sleep time in out_write() when write threshold is not reached */ @@ -138,12 +188,12 @@ #define RESAMPLER_BUFFER_FRAMES (SHORT_PERIOD_SIZE * 2) #define RESAMPLER_BUFFER_SIZE (4 * RESAMPLER_BUFFER_FRAMES) -#define DEFAULT_OUT_SAMPLING_RATE 44100 +#define DEFAULT_OUT_SAMPLING_RATE 44100 // 48000 is possible but interacts poorly with HDMI /* sampling rate when using MM low power port */ #define MM_LOW_POWER_SAMPLING_RATE 44100 /* sampling rate when using MM full power port */ -#define MM_FULL_POWER_SAMPLING_RATE 48000 +#define MM_FULL_POWER_SAMPLING_RATE 48000 // affects MULTIPLIER_FACTOR /* sampling rate when using VX port for narrow band */ #define VX_NB_SAMPLING_RATE 8000 /* sampling rate when using VX port for wide band */ @@ -225,17 +275,27 @@ enum tty_modes { struct pcm_config pcm_config_mm = { .channels = 2, .rate = MM_FULL_POWER_SAMPLING_RATE, +#ifdef PLAYBACK_MMAP .period_size = LONG_PERIOD_SIZE, .period_count = PLAYBACK_LONG_PERIOD_COUNT, +#else + .period_size = SHORT_PERIOD_SIZE, + .period_count = PLAYBACK_SHORT_PERIOD_COUNT, +#endif .format = PCM_FORMAT_S16_LE, +#ifdef PLAYBACK_MMAP .start_threshold = SHORT_PERIOD_SIZE * 2, .avail_min = LONG_PERIOD_SIZE, +#else + .start_threshold = 0, + .avail_min = 0, +#endif }; struct pcm_config pcm_config_mm_ul = { .channels = 2, .rate = MM_FULL_POWER_SAMPLING_RATE, - .period_size = SHORT_PERIOD_SIZE, + .period_size = CAPTURE_PERIOD_SIZE, .period_count = CAPTURE_PERIOD_COUNT, .format = PCM_FORMAT_S16_LE, }; @@ -1181,7 +1241,11 @@ static void select_input_device(struct tuna_audio_device *adev) static int start_output_stream(struct tuna_stream_out *out) { struct tuna_audio_device *adev = out->dev; +#ifdef PLAYBACK_MMAP unsigned int flags = PCM_OUT | PCM_MMAP | PCM_NOIRQ; +#else + unsigned int flags = PCM_OUT; +#endif int i; bool success = true; @@ -1589,6 +1653,7 @@ static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, low_power = adev->low_power && !adev->active_input; pthread_mutex_unlock(&adev->lock); +#ifdef PLAYBACK_MMAP if (low_power != out->low_power) { if (low_power) { out->write_threshold = LONG_PERIOD_SIZE * PLAYBACK_LONG_PERIOD_COUNT; @@ -1600,6 +1665,7 @@ static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, } out->low_power = low_power; } +#endif for (i = 0; i < PCM_TOTAL; i++) { if (out->pcm[i]) { @@ -1634,6 +1700,7 @@ static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, out->echo_reference->write(out->echo_reference, &b); } +#ifdef PLAYBACK_MMAP /* do not allow more than out->write_threshold frames in kernel pcm driver buffer */ do { struct timespec time_stamp; @@ -1651,16 +1718,17 @@ static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, usleep(time); } } while (kernel_frames > out->write_threshold); +#endif /* Write to all active PCMs */ for (i = 0; i < PCM_TOTAL; i++) { if (out->pcm[i]) { if (out->config[i].rate == DEFAULT_OUT_SAMPLING_RATE) { /* PCM uses native sample rate */ - ret = pcm_mmap_write(out->pcm[i], (void *)buffer, bytes); + ret = PCM_WRITE(out->pcm[i], (void *)buffer, bytes); } else { /* PCM needs resampler */ - ret = pcm_mmap_write(out->pcm[i], (void *)out->buffer, out_frames * frame_size); + ret = PCM_WRITE(out->pcm[i], (void *)out->buffer, out_frames * frame_size); } if (ret) break; |